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机场推流服务

pull/1/head
chenyukun 1 년 전
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커밋
eb9cb4f043
100개의 변경된 파일238233개의 추가작업 그리고 0개의 파일을 삭제
  1. +8
    -0
      .idea/.gitignore
  2. +6
    -0
      .idea/misc.xml
  3. +8
    -0
      .idea/modules.xml
  4. +9
    -0
      .idea/tuoheng_airport_stream.iml
  5. +6
    -0
      .idea/vcs.xml
  6. +40
    -0
      airportStream.py
  7. +30
    -0
      bean/Feedback.py
  8. +24
    -0
      bean/Result.py
  9. +20
    -0
      bean/Stream.py
  10. +0
    -0
      bean/__init__.py
  11. +17
    -0
      common/Constant.py
  12. +0
    -0
      common/__init__.py
  13. +0
    -0
      concurrency/__init__.py
  14. +62
    -0
      concurrency/http/HttpFeedbackThread.py
  15. +124
    -0
      concurrency/http/HttpPushStreamProcess.py
  16. +93
    -0
      concurrency/http/HttpServiceImpl.py
  17. +0
    -0
      concurrency/http/__init__.py
  18. +35
    -0
      concurrency/mqtt/MqttFeedbackThread.py
  19. +204
    -0
      concurrency/mqtt/MqttPushStreamProcess.py
  20. +0
    -0
      concurrency/mqtt/__init__.py
  21. +20
    -0
      config/logger.yml
  22. +20
    -0
      config/mqtt.yml
  23. +11
    -0
      config/service.yml
  24. +21
    -0
      enums/ExceptionEnum.py
  25. +9
    -0
      enums/HttpExceptionEnum.py
  26. +20
    -0
      enums/StatusEnum.py
  27. +0
    -0
      enums/__init__.py
  28. +19
    -0
      exception/CustomerException.py
  29. +0
    -0
      exception/__init__.py
  30. +674
    -0
      ffmpeg/LICENSE.txt
  31. BIN
      ffmpeg/bin/avcodec-60.dll
  32. BIN
      ffmpeg/bin/avdevice-60.dll
  33. BIN
      ffmpeg/bin/avfilter-9.dll
  34. BIN
      ffmpeg/bin/avformat-60.dll
  35. BIN
      ffmpeg/bin/avutil-58.dll
  36. BIN
      ffmpeg/bin/ffmpeg.exe
  37. BIN
      ffmpeg/bin/ffplay.exe
  38. BIN
      ffmpeg/bin/ffprobe.exe
  39. BIN
      ffmpeg/bin/postproc-57.dll
  40. BIN
      ffmpeg/bin/swresample-4.dll
  41. BIN
      ffmpeg/bin/swscale-7.dll
  42. +5
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      ffmpeg/doc/bootstrap.min.css
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      ffmpeg/doc/community.html
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      ffmpeg/doc/default.css
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      ffmpeg/doc/developer.html
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      ffmpeg/doc/faq.html
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      ffmpeg/doc/fate.html
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      ffmpeg/doc/ffmpeg-devices.html
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      ffmpeg/doc/general.html
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      ffmpeg/doc/git-howto.html
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      ffmpeg/doc/libavcodec.html
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      ffmpeg/doc/libavdevice.html
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      ffmpeg/doc/libavutil.html
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      ffmpeg/doc/libswscale.html
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      ffmpeg/doc/mailing-list-faq.html
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      ffmpeg/doc/platform.html
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      ffmpeg/doc/style.min.css
  76. +36
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      ffmpeg/include/libavcodec/ac3_parser.h
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      ffmpeg/include/libavcodec/adts_parser.h
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      ffmpeg/include/libavcodec/avcodec.h
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      ffmpeg/include/libavcodec/codec_desc.h
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      ffmpeg/include/libavcodec/codec_id.h
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      ffmpeg/include/libavcodec/codec_par.h
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      ffmpeg/include/libavcodec/dirac.h
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      ffmpeg/include/libavcodec/dxva2.h
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      ffmpeg/include/libavcodec/jni.h
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      ffmpeg/include/libavcodec/mediacodec.h
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      ffmpeg/include/libavcodec/qsv.h
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      ffmpeg/include/libavcodec/vdpau.h
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      ffmpeg/include/libavcodec/version.h
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      ffmpeg/include/libavcodec/version_major.h
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      ffmpeg/include/libavcodec/videotoolbox.h
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      ffmpeg/include/libavcodec/vorbis_parser.h
  100. +0
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      ffmpeg/include/libavcodec/xvmc.h

+ 8
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.idea/.gitignore 파일 보기

@@ -0,0 +1,8 @@
# 默认忽略的文件
/shelf/
/workspace.xml
# 基于编辑器的 HTTP 客户端请求
/httpRequests/
# Datasource local storage ignored files
/dataSources/
/dataSources.local.xml

+ 6
- 0
.idea/misc.xml 파일 보기

@@ -0,0 +1,6 @@
<?xml version="1.0" encoding="UTF-8"?>
<project version="4">
<component name="ProjectRootManager" version="2" languageLevel="JDK_19" project-jdk-name="Python 3.8 (test)" project-jdk-type="Python SDK">
<output url="file://$PROJECT_DIR$/out" />
</component>
</project>

+ 8
- 0
.idea/modules.xml 파일 보기

@@ -0,0 +1,8 @@
<?xml version="1.0" encoding="UTF-8"?>
<project version="4">
<component name="ProjectModuleManager">
<modules>
<module fileurl="file://$PROJECT_DIR$/.idea/tuoheng_airport_stream.iml" filepath="$PROJECT_DIR$/.idea/tuoheng_airport_stream.iml" />
</modules>
</component>
</project>

+ 9
- 0
.idea/tuoheng_airport_stream.iml 파일 보기

@@ -0,0 +1,9 @@
<?xml version="1.0" encoding="UTF-8"?>
<module type="JAVA_MODULE" version="4">
<component name="NewModuleRootManager" inherit-compiler-output="true">
<exclude-output />
<content url="file://$MODULE_DIR$" />
<orderEntry type="inheritedJdk" />
<orderEntry type="sourceFolder" forTests="false" />
</component>
</module>

+ 6
- 0
.idea/vcs.xml 파일 보기

@@ -0,0 +1,6 @@
<?xml version="1.0" encoding="UTF-8"?>
<project version="4">
<component name="VcsDirectoryMappings">
<mapping directory="" vcs="Git" />
</component>
</project>

+ 40
- 0
airportStream.py 파일 보기

@@ -0,0 +1,40 @@
# -*- coding: utf-8 -*-
from multiprocessing import freeze_support
from os.path import dirname, realpath, join
from loguru import logger

from service.MqttDisService import MqttDispatcherService
from service.HttpDisService import HttpDispatcherService
from util.LogUtils import init_log
from util.RWUtils import getConfigs

'''
主程序入口
'''

if __name__ == '__main__':
freeze_support()
base_dir = dirname(realpath(__file__))
init_log(base_dir)
logger.info("(♥◠‿◠)ノ゙ 【机场推流服务】开始启动 ლ(´ڡ`ლ)゙")
print("(♥◠‿◠)ノ゙ 【机场推流服务】开始启动 ლ(´ڡ`ლ)゙")
print("############################################################")
print("配置文件路径: ", join(base_dir, "config"))
print("服务配置文件: service.yml")
print("mqtt配置文件: mqtt.yml")
print("日志配置文件: logger.yml")
print("阿里云配置文件: aliyun.yml")
print("日志文件路径: ", join(base_dir, "logs"))
print("############################################################")
# mqtt交互
service_config = getConfigs(base_dir, "config/service.yml")
service_config['base_dir'] = base_dir
if 1 == service_config['docking_method']:
print("当前使用的交互模式是mqtt交互!!")
print("############################################################")
MqttDispatcherService(service_config)
elif 2 == service_config['docking_method']:
print("当前使用的交互模式是接口交互!!")
print("############################################################")
HttpDispatcherService(service_config)


+ 30
- 0
bean/Feedback.py 파일 보기

@@ -0,0 +1,30 @@
# -*-coding:utf-8 -*-
from enums.StatusEnum import StatusType
from util.QueUtil import put_queue
from util.TimeUtils import now_date_to_str


def push_result(fb_queue, errorCode="", errorMsg="", status=StatusType.RUNNING.value[0]):
put_queue(fb_queue, ('stream',
{
"errorCode": errorCode,
"errorMsg": errorMsg,
"status": status,
"currentTime": now_date_to_str()
}),
timeout=2)


def push_http_result(fb_queue, callback_url=None, errorCode="", errorMsg="", status=StatusType.RUNNING.value[0]):
if callback_url is not None:
put_queue(fb_queue, ('stream',
{
"callback_url": callback_url,
"data": {
"errorCode": errorCode,
"errorMsg": errorMsg,
"status": status,
"currentTime": now_date_to_str()
}
}),
timeout=2)

+ 24
- 0
bean/Result.py 파일 보기

@@ -0,0 +1,24 @@
# -*-coding:utf-8 -*-
from pydantic import BaseModel


class JsonResult(BaseModel):
code: int
msg: str
data: None

@staticmethod
def success(code=0, msg="操作成功!", data=None):
return {
"code": code,
"msg": msg,
"data": data
}

@staticmethod
def error(code=-1, msg="操作失败!", data=None):
return {
"code": code,
"msg": msg,
"data": data
}

+ 20
- 0
bean/Stream.py 파일 보기

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# -*-coding:utf-8 -*-
from typing import Union

from pydantic import BaseModel, Field, HttpUrl
from fastapi._compat import Required


class PushStreamRequest(BaseModel):
pullUrl: Union[str, None] = Field(default=None, title="拉流地址",
pattern="(^(https|http|rtsp|rtmp|artc|webrtc|ws)://[\\w\\d\\.\\-/:_?=&!~*'()+$@,;\"%\\[\\]]+$)?")
pushUrl: Union[str, None] = Field(default=None, title="推流地址",
pattern="(^(https|http|rtsp|rtmp|artc|webrtc|ws)://[\\w\\d\\.\\-/:_?=&!~*'()+$@,;\"%\\[\\]]+$)?")
callbackUrl: HttpUrl = Field(default=Required, title="回调地址")


class CallbackRequest(BaseModel):
errorCode: Union[str, None]
errorMsg: Union[str, None]
status: Union[int, None]
currentTime: Union[str, None]

+ 0
- 0
bean/__init__.py 파일 보기


+ 17
- 0
common/Constant.py 파일 보기

@@ -0,0 +1,17 @@
# -*- coding: utf-8 -*-
from multiprocessing import Queue

SHARE_QUEUE = Queue()


def get_share_queue():
return SHARE_QUEUE


TASK_RECORD = {
"stream": None
}


def get_task_record():
return TASK_RECORD

+ 0
- 0
common/__init__.py 파일 보기


+ 0
- 0
concurrency/__init__.py 파일 보기


+ 62
- 0
concurrency/http/HttpFeedbackThread.py 파일 보기

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# -*- coding: utf-8 -*-
import time
from json import loads
from threading import Thread
from traceback import format_exc

from loguru import logger

from util.QueUtil import get_block_queue
from util.RequestUtil import HttpRequests


class HttpFeedbackThread(Thread):
__slots__ = '__fb_queue'

def __init__(self, fb_queue):
super().__init__()
self.__fb_queue = fb_queue

def run(self):
logger.info("启动反馈线程")
fb_queue = self.__fb_queue
request = HttpRequests()
try:
headers = {'content-type': "application/json"}
while True:
try:
fb = get_block_queue(fb_queue)
if fb is not None and len(fb) > 0:
if fb[0] == "stream":
stream_req(request, fb[1], headers)
del fb
else:
time.sleep(1)
except Exception:
logger.error("反馈异常:{}", format_exc())
if request:
request.close_session()
request = HttpRequests()
finally:
request.close_session()
logger.info("反馈线程执行完成")


def stream_req(request, stream, headers):
logger.info("开始发送推流回调请求, 回调地址: {}", stream['callback_url'])
logger.info("开始发送推流回调请求, 回调请求体: {}", stream['data'])
try:
response = request.send_request('POST', stream['callback_url'], data=stream['data'],
headers=headers, timeout=3)
if response.status_code != 200:
logger.error("推流回调请求失败! 状态码: {}, {}", response.status_code, response.__dict__)
else:
content = response.content.decode('utf-8')
if content is not None and len(content) > 0:
content = loads(content)
code = content.get('code')
if code is not None and code != 0:
logger.error("推流回调请求失败! 失败描述: {}", content.get('msg'))
except Exception:
logger.error("回调请求失败, 请求体: {}, 回调地址: {}, 异常信息: {}", stream['data'], stream['callback_url'],
format_exc())

+ 124
- 0
concurrency/http/HttpPushStreamProcess.py 파일 보기

@@ -0,0 +1,124 @@
# -*- coding: utf-8 -*-
import time
from multiprocessing import Process, Queue
from threading import Thread
from traceback import format_exc

from loguru import logger

from bean.Feedback import push_http_result
from enums.ExceptionEnum import StreamStrExceptionType
from enums.StatusEnum import StatusType
from exception.CustomerException import ServiceException
from util.LogUtils import init_log
from util.PushStreamUtils import PushStreamUtil
from util.QueUtil import put_queue, get_no_block_queue


class PushStreamProcess(Process):
__slots__ = ('__fbQueue', '__event', '__service_config', '__push_stream_tool', '__callback_url')

def __init__(self, fb_queue, service_config, callback_url, pull_url, push_url):
super().__init__()
self.__fb_queue = fb_queue
self.__event = Queue()
self.__service_config = service_config
self.__callback_url = callback_url
self.__push_stream_tool = PushStreamUtil(service_config["stream"]["pullUrl"],
service_config["stream"]["pushUrl"])
self.__push_stream_tool.set_url(pull_url, push_url)

def send_event(self, result):
put_queue(self.__event, result, timeout=2, is_throw_ex=True)

def push_stream(self, push_queue, push_stream_tool):
logger.info("开始启动推流线程!")
while True:
try:
push_stream_tool.start_push_stream()
out, err = push_stream_tool.push_stream_sp.communicate()
if push_stream_tool.status:
logger.warning("推流异常,请检测拉流地址和推流地址是否正常!")
if push_stream_tool.push_stream_sp.returncode != 0:
logger.error("推流异常:{}", err.decode('utf-8'))
put_queue(push_queue, (2, StatusType.RETRYING.value[0]), timeout=2, is_throw_ex=True)
push_stream_tool.close_push_stream_p()
time.sleep(3)
if not push_stream_tool.status:
push_stream_tool.close_push_stream_p()
put_queue(push_queue, (0,), timeout=2, is_throw_ex=True)
break
except ServiceException as s:
logger.error("{}", s.msg)
push_stream_tool.close_push_stream_p()
# push_stream_tool.status = False
put_queue(push_queue, (1, s), timeout=2, is_throw_ex=True)
break
except Exception as e:
logger.error("异常:{}", format_exc())
push_stream_tool.close_push_stream_p()
# push_stream_tool.status = False
put_queue(push_queue, (1, e), timeout=2, is_throw_ex=True)
break
logger.info("推流线程运行结束!")

def run(self):
fb_queue, event_queue, callback_url = self.__fb_queue, self.__event, self.__callback_url
service_config = self.__service_config
hb_status = StatusType.WAITTING.value[0]
push_stream_tool = self.__push_stream_tool
# 初始化日志
init_log(service_config["base_dir"])
logger.info("开始启动推流进程")
push_queue = Queue()
push = Thread(target=self.push_stream, args=(push_queue, push_stream_tool))
push.setDaemon(True)
push.start()
count, start_time, ex = 0, time.time(), None
try:
while True:
if push_stream_tool.status and not push.is_alive():
logger.error("检测到推流线程异常停止!")
raise Exception("检测到推流线程异常停止!")
ph_result = get_no_block_queue(push_queue)
event_result = get_no_block_queue(event_queue)
if event_result is not None:
command = event_result.get("command")
if "stop" == command:
hb_status = StatusType.STOPPING.value[0]
push_stream_tool.status = False
push_stream_tool.close_push_stream_p()
if ph_result is not None and ph_result[0] == 0:
logger.info("推流任务停止中")
push.join(timeout=60)
push_http_result(fb_queue, callback_url, status=StatusType.SUCCESS.value[0])
break
if ph_result is not None and ph_result[0] == 1:
logger.info("推流任务异常停止中")
push.join(timeout=60)
raise ph_result[1]
if ph_result is not None and ph_result[0] == 2:
if StatusType.RETRYING.value[0] == ph_result[1]:
hb_status = StatusType.RETRYING.value[0]
start_time = time.time()
if time.time() - start_time > 10:
hb_status = StatusType.RUNNING.value[0]
count += 1
if count % 15 == 0:
push_http_result(fb_queue, callback_url, status=hb_status)
count = 0
time.sleep(1)
except ServiceException as s:
logger.error("推流异常, code: {}, msg: {}", s.code, s.msg)
ex = s.code, s.msg
except Exception:
logger.error("推流异常:{}", format_exc())
ex = StreamStrExceptionType.SERVICE_INNER_EXCEPTION.value[0], StreamStrExceptionType.SERVICE_INNER_EXCEPTION.value[1]
finally:
push_stream_tool.status = False
push_stream_tool.close_push_stream_p()
push.join(timeout=60)
if ex:
code, msg = ex
push_http_result(fb_queue, callback_url, code, msg, status=StatusType.FAILED.value[0])
logger.info("推流检测线程执行完成")

+ 93
- 0
concurrency/http/HttpServiceImpl.py 파일 보기

@@ -0,0 +1,93 @@
# -*- coding: utf-8 -*-
import sys
from multiprocessing import Queue
from threading import Thread
from time import sleep
from traceback import format_exc

from loguru import logger

from bean.Feedback import push_http_result
from common.Constant import get_task_record, get_share_queue
from concurrency.http.HttpFeedbackThread import HttpFeedbackThread
from concurrency.http.HttpPushStreamProcess import PushStreamProcess
from enums.ExceptionEnum import StreamStrExceptionType
from enums.StatusEnum import StatusType
from exception.CustomerException import ServiceException
from util.QueUtil import get_no_block_queue


class HttpServiceImpl:
__slots__ = ()

def __init__(self, service_config):
service_thread = Thread(target=self.start_service, args=(service_config,))
service_thread.setDaemon(True)
service_thread.start()

@staticmethod
def start_service(service_config):
fb_queue = Queue()
task, msg_queue = get_task_record(), get_share_queue()
handle_method = {
"stream": lambda x, y, z, h: handle_stream(x, y, z, h)
}
feedbackThread = None
while True:
try:
if task["stream"] is not None and not task["stream"].is_alive():
task["stream"] = None
feedbackThread = start_feedback_thread(feedbackThread, fb_queue)
message = get_no_block_queue(msg_queue)
if message is not None and isinstance(message, tuple) and len(message) == 2:
if handle_method.get(message[0]) is not None:
handle_method[message[0]](message[1], service_config, task, fb_queue)
else:
sleep(1)
except Exception:
logger.error("服务异常: {}", format_exc())


def handle_stream(msg, service_config, task, fb_queue):
try:
command = msg["command"]
if 'start' == command:
if task["stream"] is not None:
logger.error("推流任务已存在!!!")
push_http_result(fb_queue, msg["callback_url"],
StreamStrExceptionType.PUSH_STREAM_TASK_IS_AREADLY.value[0],
StreamStrExceptionType.PUSH_STREAM_TASK_IS_AREADLY.value[1],
StatusType.FAILED.value[0])
return
push_http_result(fb_queue, msg["callback_url"], status=StatusType.WAITTING.value[0])
pp = PushStreamProcess(fb_queue, service_config, msg["callback_url"], msg["pull_url"], msg["push_url"])
pp.start()
task["stream"] = pp
elif 'stop' == command:
if task["stream"] is None:
logger.error("推流任务不存在, 任务无法停止!")
return
task["stream"].send_event({"command": "stop"})
except ServiceException as s:
logger.error("消息处理异常: {}", s.msg)
push_http_result(fb_queue, msg["callback_url"], s.code, s.msg, StatusType.FAILED.value[0])
except Exception:
logger.error("消息处理异常: {}", format_exc())
push_http_result(fb_queue, msg["callback_url"],
StreamStrExceptionType.SERVICE_INNER_EXCEPTION.value[0],
StreamStrExceptionType.SERVICE_INNER_EXCEPTION.value[1],
StatusType.FAILED.value[0])
finally:
del msg


def start_feedback_thread(feedbackThread, fb_queue):
if feedbackThread is None:
feedbackThread = HttpFeedbackThread(fb_queue)
feedbackThread.setDaemon(True)
feedbackThread.start()
else:
if not feedbackThread.is_alive():
logger.error("反馈线程异常停止! 开始终止程序!")
sys.exit()
return feedbackThread

+ 0
- 0
concurrency/http/__init__.py 파일 보기


+ 35
- 0
concurrency/mqtt/MqttFeedbackThread.py 파일 보기

@@ -0,0 +1,35 @@
# -*- coding: utf-8 -*-
import time
from threading import Thread
from traceback import format_exc

from loguru import logger

from util.QueUtil import get_block_queue

'''
问题反馈线程
'''


class FeedbackThread(Thread):
__slots__ = ('__fb_queue', '__mq')

def __init__(self, fb_queue, mq):
super().__init__()
self.__fb_queue = fb_queue
self.__mq = mq

def run(self):
logger.info("启动反馈线程")
fb_queue, mq = self.__fb_queue, self.__mq
while True:
try:
fb = get_block_queue(fb_queue)
if fb is not None and len(fb) > 0:
mq.publish(fb)
else:
time.sleep(1)
except Exception:
logger.error("反馈异常:{}", format_exc())
logger.info("反馈线程执行完成")

+ 204
- 0
concurrency/mqtt/MqttPushStreamProcess.py 파일 보기

@@ -0,0 +1,204 @@
# -*- coding: utf-8 -*-
import time
from multiprocessing import Queue, Process
from threading import Thread
from traceback import format_exc

from loguru import logger

from bean.Feedback import push_result
from enums.ExceptionEnum import ExceptionType
from enums.StatusEnum import StatusType
from exception.CustomerException import ServiceException
from util.LogUtils import init_log
from util.PushStreamUtils import PushStreamUtil
from util.QueUtil import put_queue, get_no_block_queue


class PushStreamProcess(Process):
__slots__ = ('__fb_queue', '__service_config', '__event', '__pullUrl', '__pushUrl')

def __init__(self, fb_queue, service_config, pullUrl, pushUrl):
super().__init__()
self.__fb_queue = fb_queue
self.__service_config = service_config
self.__event = Queue()
self.__pullUrl = pullUrl
self.__pushUrl = pushUrl
push_result(fb_queue, status=StatusType.WAITTING.value[0])

def send_event(self, result):
put_queue(self.__event, result, timeout=2, is_throw_ex=True)

@staticmethod
def push_stream(push_queue, push_stream_tool):
logger.info("开始启动推流线程!")
while True:
try:
push_stream_tool.start_push_stream()
out, err = push_stream_tool.push_stream_sp.communicate()
if push_stream_tool.status:
logger.warning("推流异常,请检测拉流地址和推流地址是否正常!")
if push_stream_tool.push_stream_sp.returncode != 0:
logger.error("推流异常:{}", err.decode("utf-8"))
put_queue(push_queue, (2, StatusType.RETRYING.value[0]), timeout=2, is_throw_ex=True)
push_stream_tool.close_push_stream_p()
time.sleep(1)
if not push_stream_tool.status:
push_stream_tool.close_push_stream_p()
put_queue(push_queue, (0,), timeout=2, is_throw_ex=True)
break
except ServiceException as s:
logger.error("异常: {}", s.msg)
push_stream_tool.close_push_stream_p()
# push_stream_tool.status = False
put_queue(push_queue, (1, s), timeout=2, is_throw_ex=True)
break
except Exception as e:
logger.error("异常: {}", format_exc())
push_stream_tool.close_push_stream_p()
# push_stream_tool.status = False
put_queue(push_queue, (1, e), timeout=2, is_throw_ex=True)
break
logger.info("推流线程运行结束!")

def run(self):
service_config, pullUrl, pushUrl = self.__service_config, self.__pullUrl, self.__pushUrl
fb_queue, event_queue, push_queue = self.__fb_queue, self.__event, Queue()
hb_status = StatusType.WAITTING.value[0]
push_stream_tool = PushStreamUtil(service_config["stream"]["pullUrl"], service_config["stream"]["pushUrl"])
push_stream_tool.set_url(pullUrl, pushUrl)
push = None
count, start_time, ex = 0, time.time(), None
try:
init_log(service_config["base_dir"])
logger.info("开始启动推流进程!")
push = Thread(target=self.push_stream, args=(push_queue, push_stream_tool))
push.setDaemon(True)
push.start()
while True:
if push_stream_tool.status and not push.is_alive():
logger.error("检测到推流线程异常停止!")
raise Exception("检测到推流线程异常停止!")
ph_result = get_no_block_queue(push_queue)
event_result = get_no_block_queue(event_queue)
if event_result is not None:
command = event_result.get("command")
if "stop" == command:
hb_status = StatusType.STOPPING.value[0]
push_stream_tool.status = False
push_stream_tool.close_push_stream_p()
if ph_result is not None and ph_result[0] == 0:
logger.info("推流任务停止中")
push.join(timeout=60)
push_result(fb_queue, status=StatusType.SUCCESS.value[0])
break
if ph_result is not None and ph_result[0] == 1:
logger.info("推流任务异常停止中")
push.join(timeout=60)
raise ph_result[1]
if ph_result is not None and ph_result[0] == 2:
if StatusType.RETRYING.value[0] == ph_result[1]:
hb_status = StatusType.RETRYING.value[0]
start_time = time.time()
if time.time() - start_time > 20:
hb_status = StatusType.RUNNING.value[0]
count += 1
if count % 15 == 0:
push_result(fb_queue, status=hb_status)
count = 0
time.sleep(1)
except ServiceException as s:
logger.error("推流异常, code: {}, msg: {}", s.code, s.msg)
ex = s.code, s.msg
except Exception:
logger.error("推流异常:{}", format_exc())
ex = ExceptionType.SERVICE_INNER_EXCEPTION.value[0], ExceptionType.SERVICE_INNER_EXCEPTION.value[1]
finally:
push_stream_tool.status = False
push_stream_tool.close_push_stream_p()
if push:
push.join(timeout=60)
if ex:
code, msg = ex
push_result(fb_queue, code, msg, status=StatusType.FAILED.value[0])
logger.info("推流检测线程执行完成")

# """
# 版本二
# """
# class PushStreamThread(Thread):
# __slots__ = ('__fbQueue', '__event', '__push_stream_tool', '__hb_status')
#
# def __init__(self, fbQueue, push_stream_tool):
# super().__init__()
# self.__fb_queue = fbQueue
# self.__event = Queue()
# self.__push_stream_tool = push_stream_tool
# self.__hb_status = StatusType.WAITTING.value[0]
# put_queue(self.__fb_queue, {
# "errorCode": "",
# "errorMsg": "",
# "status": StatusType.WAITTING.value[0],
# "current_time": now_date_to_str()}, is_throw_ex=False)
#
# def send_event(self, result):
# put_queue(self.__event, result, is_throw_ex=False)
#
# def run(self):
# logger.info("启动推流线程")
# self.__push_stream_tool.start_push_stream()
# count = 0
# start_time = time.time()
# while True:
# try:
# event_result = get_no_block_queue(self.__event)
# if event_result is not None:
# command = event_result.get("command")
# if "stop" == command:
# self.__push_stream_tool.close_push_stream_p(send=True)
# put_queue(self.__fb_queue, {
# "errorCode": "",
# "errorMsg": "",
# "status": StatusType.SUCCESS.value[0],
# "current_time": now_date_to_str()}, is_throw_ex=False)
# break
# if self.__push_stream_tool.push_stream_sp and self.__push_stream_tool.push_stream_sp.poll() is not None:
# logger.error("推流异常,请检查推流地址和拉流地址是否正常!!!!!")
# # if self.__push_stream_tool.push_stream_sp.returncode != 0:
# # out, err = self.__push_stream_tool.push_stream_sp.communicate(timeout=120)
# # logger.error("推流异常:{}", err.decode())
# self.__push_stream_tool.close_push_stream_p(send=True)
# self.__push_stream_tool.start_push_stream()
# self.__hb_status = StatusType.RETRYING.value[0]
# start_time = time.time()
# if time.time() - start_time > 20:
# self.__hb_status = StatusType.RUNNING.value[0]
# count += 1
# if count % 10 == 0:
# put_queue(self.__fb_queue, {
# "errorCode": "",
# "errorMsg": "",
# "status": self.__hb_status,
# "current_time": now_date_to_str()}, is_throw_ex=False)
# count = 0
# time.sleep(1)
# except ServiceException as s:
# logger.error("推流异常, code: {}, msg: {}", s.code, s.msg)
# put_queue(self.__fb_queue, {
# "errorCode": s.code,
# "errorMsg": s.msg,
# "status": StatusType.FAILED.value[0],
# "current_time": now_date_to_str()}, is_throw_ex=False)
# self.__push_stream_tool.close_push_stream_p(send=True)
# break
# except Exception:
# logger.error("推流异常:{}", format_exc())
# put_queue(self.__fb_queue, {
# "errorCode": ExceptionType.SERVICE_INNER_EXCEPTION.value[0],
# "errorMsg": ExceptionType.SERVICE_INNER_EXCEPTION.value[1],
# "status": StatusType.FAILED.value[0],
# "current_time": now_date_to_str()}, is_throw_ex=False)
# self.__push_stream_tool.close_push_stream_p(send=True)
# break
# logger.info("推流检测线程执行完成")

+ 0
- 0
concurrency/mqtt/__init__.py 파일 보기


+ 20
- 0
config/logger.yml 파일 보기

@@ -0,0 +1,20 @@
# 是否启动日志文件记录日志
enable_file_log: true
# 是否启动控制台打印日志
enable_stderr: true
# 日志相对于根路径下的相对路径
base_path: "logs"
# 日志文件名称
log_name: "airport_media.log"
# 日志打印格式
log_fmt: "{time:YYYY-MM-DD HH:mm:ss.SSS} [{level}][{process.name}-{process.id}-{thread.name}-{thread.id}][{line}] {module}-{function} - {message}"
# 日志打印级别
level: "INFO"
# 日志绕接时间
rotation: "00:00"
# 日志保存时间
retention: "7 days"
# 日志编码格式
encoding: "utf8"



+ 20
- 0
config/mqtt.yml 파일 보기

@@ -0,0 +1,20 @@
mqtt:
# mqtt客户端id对应每个机场平台的编码
client_id: "THOBS@0000THJSQ232003"
# mqtt用户(根据对接环境修改)
username: "admin"
# mqtt密码(根据对接环境修改)
password: "admin##123"
# mqtt连接地址(根据对接环境修改)
host: "mqtt.t-aaron.com"
# mqtt端口号(根据对接环境修改)
port: 10883
# 长链接时间
keepalive: 60
topic:
stream:
# 推流订阅topic(不修改)
sub_topic: "/v1/%s/stream/push"
# 推流响应topic(不修改)
res_topic: "/v1/%s/stream/result"


+ 11
- 0
config/service.yml 파일 보기

@@ -0,0 +1,11 @@
# 1: mqtt消息队列方式对接
# 2: http接口方式对接
docking_method: 1
# 推拉流功能配置
stream:
# 拉流地址
pullUrl: "rtmp://live.play.t-aaron.com/live/1111"
# 推流地址
pushUrl: "rtmp://live.push.t-aaron.com/live/2222"



+ 21
- 0
enums/ExceptionEnum.py 파일 보기

@@ -0,0 +1,21 @@
from enum import Enum, unique


@unique
class ExceptionType(Enum):
ILLEGAL_PARAMETER_FORMAT = ("CN000", "参数格式错误!")

SERVICE_INNER_EXCEPTION = ("CN999", "系统内部异常!")


@unique
class StreamStrExceptionType(Enum):
PUSH_STREAM_TASK_IS_AREADLY = ("PT000", "推流任务已存在!")

PUSH_STREAM_TASK_IS_NOT_AREADLY = ("PT001", "推流任务不存在!")

PULL_STREAM_URL_IS_NULL = ("PT002", "拉流地址不能为空!")

PUSH_STREAM_URL_IS_NULL = ("PT003", "推流地址不能为空!")

SERVICE_INNER_EXCEPTION = ("PT999", "系统内部异常!")

+ 9
- 0
enums/HttpExceptionEnum.py 파일 보기

@@ -0,0 +1,9 @@
from enum import Enum, unique


@unique
class StreamExceptionType(Enum):

TASK_IS_EXECUTING = (-1, "推流任务正在执行, 请稍后再试!")

TASK_NOT_EXISTS = (-1, "推流任务停止失败, 任务不存在!")

+ 20
- 0
enums/StatusEnum.py 파일 보기

@@ -0,0 +1,20 @@
from enum import Enum, unique


# 异常枚举
@unique
class StatusType(Enum):

WAITTING = (5, "待推流")

RETRYING = (10, "重试中")

RUNNING = (15, "推流中")

STOPPING = (20, "停止中")

SUCCESS = (25, "完成")

TIMEOUT = (30, "超时")

FAILED = (35, "失败")

+ 0
- 0
enums/__init__.py 파일 보기


+ 19
- 0
exception/CustomerException.py 파일 보기

@@ -0,0 +1,19 @@
# -*- coding: utf-8 -*-
from loguru import logger


"""
自定义异常
"""


class ServiceException(Exception):
def __init__(self, code, msg):
self.code = code
self.msg = msg

def __str__(self):
logger.error("异常编码:{}, 异常描述:{}", self.code, self.msg)




+ 0
- 0
exception/__init__.py 파일 보기


+ 674
- 0
ffmpeg/LICENSE.txt 파일 보기

@@ -0,0 +1,674 @@
GNU GENERAL PUBLIC LICENSE
Version 3, 29 June 2007

Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.

Preamble

The GNU General Public License is a free, copyleft license for
software and other kinds of works.

The licenses for most software and other practical works are designed
to take away your freedom to share and change the works. By contrast,
the GNU General Public License is intended to guarantee your freedom to
share and change all versions of a program--to make sure it remains free
software for all its users. We, the Free Software Foundation, use the
GNU General Public License for most of our software; it applies also to
any other work released this way by its authors. You can apply it to
your programs, too.

When we speak of free software, we are referring to freedom, not
price. Our General Public Licenses are designed to make sure that you
have the freedom to distribute copies of free software (and charge for
them if you wish), that you receive source code or can get it if you
want it, that you can change the software or use pieces of it in new
free programs, and that you know you can do these things.

To protect your rights, we need to prevent others from denying you
these rights or asking you to surrender the rights. Therefore, you have
certain responsibilities if you distribute copies of the software, or if
you modify it: responsibilities to respect the freedom of others.

For example, if you distribute copies of such a program, whether
gratis or for a fee, you must pass on to the recipients the same
freedoms that you received. You must make sure that they, too, receive
or can get the source code. And you must show them these terms so they
know their rights.

Developers that use the GNU GPL protect your rights with two steps:
(1) assert copyright on the software, and (2) offer you this License
giving you legal permission to copy, distribute and/or modify it.

For the developers' and authors' protection, the GPL clearly explains
that there is no warranty for this free software. For both users' and
authors' sake, the GPL requires that modified versions be marked as
changed, so that their problems will not be attributed erroneously to
authors of previous versions.

Some devices are designed to deny users access to install or run
modified versions of the software inside them, although the manufacturer
can do so. This is fundamentally incompatible with the aim of
protecting users' freedom to change the software. The systematic
pattern of such abuse occurs in the area of products for individuals to
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have designed this version of the GPL to prohibit the practice for those
products. If such problems arise substantially in other domains, we
stand ready to extend this provision to those domains in future versions
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Finally, every program is threatened constantly by software patents.
States should not allow patents to restrict development and use of
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patents cannot be used to render the program non-free.

The precise terms and conditions for copying, distribution and
modification follow.

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Nothing in this License shall be construed as excluding or limiting
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Notwithstanding any other provision of this License, you have
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END OF TERMS AND CONDITIONS

How to Apply These Terms to Your New Programs

If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it
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to attach them to the start of each source file to most effectively
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the "copyright" line and a pointer to where the full notice is found.

<one line to give the program's name and a brief idea of what it does.>
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This program is free software: you can redistribute it and/or modify
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(at your option) any later version.

This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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You should have received a copy of the GNU General Public License
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Also add information on how to contact you by electronic and paper mail.

If the program does terminal interaction, make it output a short
notice like this when it starts in an interactive mode:

<program> Copyright (C) <year> <name of author>
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under certain conditions; type `show c' for details.

The hypothetical commands `show w' and `show c' should show the appropriate
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might be different; for a GUI interface, you would use an "about box".

You should also get your employer (if you work as a programmer) or school,
if any, to sign a "copyright disclaimer" for the program, if necessary.
For more information on this, and how to apply and follow the GNU GPL, see
<http://www.gnu.org/licenses/>.

The GNU General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications with
the library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License. But first, please read
<http://www.gnu.org/philosophy/why-not-lgpl.html>.

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Community
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Community
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Organisation-1" href="#Organisation-1">1 Organisation</a></li>
<li><a id="toc-General-Assembly-1" href="#General-Assembly-1">2 General Assembly</a></li>
<li><a id="toc-Voting-1" href="#Voting-1">3 Voting</a></li>
<li><a id="toc-Technical-Committee-1" href="#Technical-Committee-1">4 Technical Committee</a>
<ul class="no-bullet">
<li><a id="toc-Resolution-Process-1" href="#Resolution-Process-1">4.1 Resolution Process</a>
<ul class="no-bullet">
<li><a id="toc-Seizing" href="#Seizing">4.1.1 Seizing</a></li>
<li><a id="toc-Announcement" href="#Announcement">4.1.2 Announcement</a></li>
<li><a id="toc-RFC-call" href="#RFC-call">4.1.3 RFC call</a></li>
<li><a id="toc-Within-TC" href="#Within-TC">4.1.4 Within TC</a></li>
<li><a id="toc-Decisions" href="#Decisions">4.1.5 Decisions</a></li>
</ul></li>
</ul></li>
<li><a id="toc-Community-Committee-1" href="#Community-Committee-1">5 Community Committee</a></li>
<li><a id="toc-Code-of-Conduct-1" href="#Code-of-Conduct-1">6 Code of Conduct</a></li>
</ul>
</div>
</div>

<span id="Organisation"></span><a name="Organisation-1"></a>
<h2 class="chapter">1 Organisation<span class="pull-right"><a class="anchor hidden-xs" href="#Organisation-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Organisation-1" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg project is organized through a community working on global consensus.
</p>
<p>Decisions are taken by the ensemble of active members, through voting and are aided by two committees.
</p>
<span id="General-Assembly"></span><a name="General-Assembly-1"></a>
<h2 class="chapter">2 General Assembly<span class="pull-right"><a class="anchor hidden-xs" href="#General-Assembly-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-General-Assembly-1" aria-hidden="true">TOC</a></span></h2>

<p>The ensemble of active members is called the General Assembly (GA).
</p>
<p>The General Assembly is sovereign and legitimate for all its decisions regarding the FFmpeg project.
</p>
<p>The General Assembly is made up of active contributors.
</p>
<p>Contributors are considered &quot;active contributors&quot; if they have pushed more than 20 patches in the last 36 months in the main FFmpeg repository, or if they have been voted in by the GA.
</p>
<p>Additional members are added to the General Assembly through a vote after proposal by a member of the General Assembly. They are part of the GA for two years, after which they need a confirmation by the GA.
</p>
<p>A script to generate the current members of the general assembly (minus members voted in) can be found in &lsquo;tools/general_assembly.pl&lsquo;.
</p>
<span id="Voting"></span><a name="Voting-1"></a>
<h2 class="chapter">3 Voting<span class="pull-right"><a class="anchor hidden-xs" href="#Voting-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Voting-1" aria-hidden="true">TOC</a></span></h2>

<p>Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
</p>
<p>Majority vote means more than 50% of the expressed ballots.
</p>
<span id="Technical-Committee"></span><a name="Technical-Committee-1"></a>
<h2 class="chapter">4 Technical Committee<span class="pull-right"><a class="anchor hidden-xs" href="#Technical-Committee-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Technical-Committee-1" aria-hidden="true">TOC</a></span></h2>

<p>The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project. They will consider the merits of all the positions, judge them and make a decision.
</p>
<p>The TC resolves technical conflicts but is not a technical steering committee.
</p>
<p>Decisions by the TC are binding for all the contributors.
</p>
<p>Decisions made by the TC can be re-opened after 1 year or by a majority vote of the General Assembly, requested by one of the member of the GA.
</p>
<p>The TC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the TC.
</p>
<p>The members of the TC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
</p>
<p>The conflict resolution process is detailed in the resolution process document.
</p>
<p>The TC can be contacted at &lt;tc@ffmpeg&gt;.
</p>
<span id="Resolution-Process"></span><a name="Resolution-Process-1"></a>
<h3 class="section">4.1 Resolution Process<span class="pull-right"><a class="anchor hidden-xs" href="#Resolution-Process-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Resolution-Process-1" aria-hidden="true">TOC</a></span></h3>

<p>The Technical Committee (TC) is here to arbitrate and make decisions when technical conflicts occur in the project.
</p>
<p>The TC main role is to resolve technical conflicts. It is therefore not a technical steering committee, but it is understood that some decisions might impact the future of the project.
</p>
<a name="Seizing"></a>
<h4 class="subsection">4.1.1 Seizing<span class="pull-right"><a class="anchor hidden-xs" href="#Seizing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Seizing" aria-hidden="true">TOC</a></span></h4>

<p>The TC can take possession of any technical matter that it sees fit.
</p>
<p>To involve the TC in a matter, email tc&nbsp;or CC them on an ongoing discussion.
</p>
<p>As members of TC are developers, they also can email tc&nbsp;to raise an issue.
</p><a name="Announcement"></a>
<h4 class="subsection">4.1.2 Announcement<span class="pull-right"><a class="anchor hidden-xs" href="#Announcement" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Announcement" aria-hidden="true">TOC</a></span></h4>

<p>The TC, once seized, must announce itself on the main mailing list, with a [TC] tag.
</p>
<p>The TC has 2 modes of operation: a RFC one and an internal one.
</p>
<p>If the TC thinks it needs the input from the larger community, the TC can call for a RFC. Else, it can decide by itself.
</p>
<p>If the disagreement involves a member of the TC, that member should recuse themselves from the decision.
</p>
<p>The decision to use a RFC process or an internal discussion is a discretionary decision of the TC.
</p>
<p>The TC can also reject a seizure for a few reasons such as: the matter was not discussed enough previously; it lacks expertise to reach a beneficial decision on the matter; or the matter is too trivial.
</p><a name="RFC-call"></a>
<h4 class="subsection">4.1.3 RFC call<span class="pull-right"><a class="anchor hidden-xs" href="#RFC-call" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-RFC-call" aria-hidden="true">TOC</a></span></h4>

<p>In the RFC mode, one person from the TC posts on the mailing list the technical question and will request input from the community.
</p>
<p>The mail will have the following specification:
</p>
<p>a precise title
a specific tag [TC RFC]
a top-level email
contain a precise question that does not exceed 100 words and that is answerable by developers
may have an extra description, or a link to a previous discussion, if deemed necessary,
contain a precise end date for the answers.
</p>
<p>The answers from the community must be on the main mailing list and must have the following specification:
</p>
<p>keep the tag and the title unchanged
limited to 400 words
a first-level, answering directly to the main email
answering to the question.
</p>
<p>Further replies to answers are permitted, as long as they conform to the community standards of politeness, they are limited to 100 words, and are not nested more than once. (max-depth=2)
</p>
<p>After the end-date, mails on the thread will be ignored.
</p>
<p>Violations of those rules will be escalated through the Community Committee.
</p>
<p>After all the emails are in, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay, that will be notified on the mailing list.
</p><a name="Within-TC"></a>
<h4 class="subsection">4.1.4 Within TC<span class="pull-right"><a class="anchor hidden-xs" href="#Within-TC" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Within-TC" aria-hidden="true">TOC</a></span></h4>

<p>In the internal case, the TC has 96 hours to give its final decision. Exceptionally, the TC can request an extra delay.
</p><a name="Decisions"></a>
<h4 class="subsection">4.1.5 Decisions<span class="pull-right"><a class="anchor hidden-xs" href="#Decisions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Decisions" aria-hidden="true">TOC</a></span></h4>

<p>The decisions from the TC will be sent on the mailing list, with the [TC] tag.
</p>
<p>Internally, the TC should take decisions with a majority, or using ranked-choice voting.
</p>
<p>The decision from the TC should be published with a summary of the reasons that lead to this decision.
</p>
<p>The decisions from the TC are final, until the matters are reopened after no less than one year.
</p>
<span id="Community-Committee"></span><a name="Community-Committee-1"></a>
<h2 class="chapter">5 Community Committee<span class="pull-right"><a class="anchor hidden-xs" href="#Community-Committee-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Community-Committee-1" aria-hidden="true">TOC</a></span></h2>

<p>The Community Committee (CC) is here to arbitrage and make decisions when inter-personal conflicts occur in the project. It will decide quickly and take actions, for the sake of the project.
</p>
<p>The CC can remove privileges of offending members, including removal of commit access and temporary ban from the community.
</p>
<p>Decisions made by the CC can be re-opened after 1 year or by a majority vote of the General Assembly. Indefinite bans from the community must be confirmed by the General Assembly, in a majority vote.
</p>
<p>The CC is elected by the General Assembly for a duration of 1 year, and is composed of 5 members. Members can be re-elected if they wish. A majority vote in the General Assembly can trigger a new election of the CC.
</p>
<p>The members of the CC can be elected from outside of the GA. Candidates for election can either be suggested or self-nominated.
</p>
<p>The CC is governed by and responsible for enforcing the Code of Conduct.
</p>
<p>The CC can be contacted at &lt;cc@ffmpeg&gt;.
</p>
<span id="Code-of-Conduct"></span><a name="Code-of-Conduct-1"></a>
<h2 class="chapter">6 Code of Conduct<span class="pull-right"><a class="anchor hidden-xs" href="#Code-of-Conduct-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code-of-Conduct-1" aria-hidden="true">TOC</a></span></h2>

<p>Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
</p>
<p>Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
</p>
<p>Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it&rsquo;s rarely good to start with that as initial assumption.
</p>
<p>Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
</p>
<p>Try to help other team members and cooperate if you can.
</p>
<p>The goal of software development is to create technical excellence, not for any
individual to be better and &quot;win&quot; against the others. Large software projects
are only possible and successful through teamwork.
</p>
<p>If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
</p>
<p>Finally, keep in mind the immortal words of Bill and Ted,
&quot;Be excellent to each other.&quot;
</p>
<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Developer Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Developer Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Introduction" href="#Introduction">1 Introduction</a>
<ul class="no-bullet">
<li><a id="toc-Contributing-code" href="#Contributing-code">1.1 Contributing code</a></li>
</ul></li>
<li><a id="toc-Coding-Rules-1" href="#Coding-Rules-1">2 Coding Rules</a>
<ul class="no-bullet">
<li><a id="toc-C-language-features" href="#C-language-features">2.1 C language features</a></li>
<li><a id="toc-Code-formatting-conventions" href="#Code-formatting-conventions">2.2 Code formatting conventions</a>
<ul class="no-bullet">
<li><a id="toc-Vim-configuration" href="#Vim-configuration">2.2.1 Vim configuration</a></li>
<li><a id="toc-Emacs-configuration" href="#Emacs-configuration">2.2.2 Emacs configuration</a></li>
</ul></li>
<li><a id="toc-Comments" href="#Comments">2.3 Comments</a></li>
<li><a id="toc-Naming-conventions" href="#Naming-conventions">2.4 Naming conventions</a></li>
<li><a id="toc-Miscellaneous-conventions" href="#Miscellaneous-conventions">2.5 Miscellaneous conventions</a></li>
</ul></li>
<li><a id="toc-Development-Policy-1" href="#Development-Policy-1">3 Development Policy</a>
<ul class="no-bullet">
<li><a id="toc-Patches_002fCommitting" href="#Patches_002fCommitting">3.1 Patches/Committing</a></li>
<li><a id="toc-Code" href="#Code">3.2 Code</a></li>
<li><a id="toc-Documentation_002fOther" href="#Documentation_002fOther">3.3 Documentation/Other</a></li>
</ul></li>
<li><a id="toc-Submitting-patches-1" href="#Submitting-patches-1">4 Submitting patches</a></li>
<li><a id="toc-New-codecs-or-formats-checklist" href="#New-codecs-or-formats-checklist">5 New codecs or formats checklist</a></li>
<li><a id="toc-Patch-submission-checklist" href="#Patch-submission-checklist">6 Patch submission checklist</a></li>
<li><a id="toc-Patch-review-process" href="#Patch-review-process">7 Patch review process</a></li>
<li><a id="toc-Regression-tests-1" href="#Regression-tests-1">8 Regression tests</a>
<ul class="no-bullet">
<li><a id="toc-Adding-files-to-the-fate_002dsuite-dataset" href="#Adding-files-to-the-fate_002dsuite-dataset">8.1 Adding files to the fate-suite dataset</a></li>
<li><a id="toc-Visualizing-Test-Coverage" href="#Visualizing-Test-Coverage">8.2 Visualizing Test Coverage</a></li>
<li><a id="toc-Using-Valgrind" href="#Using-Valgrind">8.3 Using Valgrind</a></li>
</ul></li>
<li><a id="toc-Release-process-1" href="#Release-process-1">9 Release process</a>
<ul class="no-bullet">
<li><a id="toc-Criteria-for-Point-Releases-1" href="#Criteria-for-Point-Releases-1">9.1 Criteria for Point Releases</a></li>
<li><a id="toc-Release-Checklist" href="#Release-Checklist">9.2 Release Checklist</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="Introduction"></a>
<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>

<p>This text is concerned with the development <em>of</em> FFmpeg itself. Information
on using the FFmpeg libraries in other programs can be found elsewhere, e.g. in:
</p><ul>
<li> the installed header files
</li><li> <a href="http://ffmpeg.org/doxygen/trunk/index.html">the Doxygen documentation</a>
generated from the headers
</li><li> the examples under <samp>doc/examples</samp>
</li></ul>

<p>If you modify FFmpeg code for your own use case, you are highly encouraged to
<em>submit your changes back to us</em>, using this document as a guide. There are
both pragmatic and ideological reasons to do so:
</p><ul>
<li> Maintaining external changes to keep up with upstream development is
time-consuming and error-prone. With your code in the main tree, it will be
maintained by FFmpeg developers.
</li><li> FFmpeg developers include leading experts in the field who can find bugs or
design flaws in your code.
</li><li> By supporting the project you find useful you ensure it continues to be
maintained and developed.
</li></ul>

<p>For more detailed legal information about the use of FFmpeg in
external programs read the <samp>LICENSE</samp> file in the source tree and
consult <a href="https://ffmpeg.org/legal.html">https://ffmpeg.org/legal.html</a>.
</p>
<a name="Contributing-code"></a>
<h3 class="section">1.1 Contributing code<span class="pull-right"><a class="anchor hidden-xs" href="#Contributing-code" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Contributing-code" aria-hidden="true">TOC</a></span></h3>

<p>All proposed code changes should be submitted for review to
<a href="mailto:ffmpeg-devel@ffmpeg.org">the development mailing list</a>, as
described in more detail in the <a href="#Submitting-patches">Submitting patches</a> chapter. The code
should comply with the <a href="#Development-Policy">Development Policy</a> and follow the <a href="#Coding-Rules">Coding Rules</a>.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
</p>
<span id="Coding-Rules"></span><a name="Coding-Rules-1"></a>
<h2 class="chapter">2 Coding Rules<span class="pull-right"><a class="anchor hidden-xs" href="#Coding-Rules-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Coding-Rules-1" aria-hidden="true">TOC</a></span></h2>

<a name="C-language-features"></a>
<h3 class="section">2.1 C language features<span class="pull-right"><a class="anchor hidden-xs" href="#C-language-features" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-C-language-features" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg is programmed in the ISO C99 language, extended with:
</p><ul>
<li> Atomic operations from C11 <samp>stdatomic.h</samp>. They are emulated on
architectures/compilers that do not support them, so all FFmpeg-internal code
may use atomics without any extra checks. However, <samp>stdatomic.h</samp> must not
be included in public headers, so they stay C99-compatible.
</li></ul>

<p>Compiler-specific extensions may be used with good reason, but must not be
depended on, i.e. the code must still compile and work with compilers lacking
the extension.
</p>
<p>The following C99 features must not be used anywhere in the codebase:
</p><ul>
<li> variable-length arrays;

</li><li> complex numbers;

</li><li> mixed statements and declarations.
</li></ul>

<a name="Code-formatting-conventions"></a>
<h3 class="section">2.2 Code formatting conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Code-formatting-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code-formatting-conventions" aria-hidden="true">TOC</a></span></h3>

<p>There are the following guidelines regarding the indentation in files:
</p>
<ul>
<li> Indent size is 4.

</li><li> The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.

</li><li> You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.

</li><li> K&amp;R coding style is used.
</li></ul>
<p>The presentation is one inspired by &rsquo;indent -i4 -kr -nut&rsquo;.
</p>
<a name="Vim-configuration"></a>
<h4 class="subsection">2.2.1 Vim configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Vim-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Vim-configuration" aria-hidden="true">TOC</a></span></h4>
<p>In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your <samp>.vimrc</samp>:
</p><div class="example">
<pre class="example">&quot; indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
&quot; Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
&quot; Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
&quot; Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@&lt;!$/
</pre></div>

<a name="Emacs-configuration"></a>
<h4 class="subsection">2.2.2 Emacs configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Emacs-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Emacs-configuration" aria-hidden="true">TOC</a></span></h4>
<p>For Emacs, add these roughly equivalent lines to your <samp>.emacs.d/init.el</samp>:
</p><div class="example lisp">
<pre class="lisp">(c-add-style &quot;ffmpeg&quot;
'(&quot;k&amp;r&quot;
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style &quot;ffmpeg&quot;)
</pre></div>

<a name="Comments"></a>
<h3 class="section">2.3 Comments<span class="pull-right"><a class="anchor hidden-xs" href="#Comments" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Comments" aria-hidden="true">TOC</a></span></h3>
<p>Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
</p>
<p>Avoid Qt-style and similar Doxygen syntax with <code>!</code> in it, i.e. replace
<code>//!</code> with <code>///</code> and similar. Also @ syntax should be employed
for markup commands, i.e. use <code>@param</code> and not <code>\param</code>.
</p>
<div class="example">
<pre class="example">/**
* @file
* MPEG codec.
* @author ...
*/

/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar {
int var1; /**&lt; var1 description */
int var2; ///&lt; var2 description
/** var3 description */
int var3;
} Foobar;

/**
* Summary sentence.
* more text ...
* ...
* @param my_parameter description of my_parameter
* @return return value description
*/
int myfunc(int my_parameter)
...
</pre></div>

<a name="Naming-conventions"></a>
<h3 class="section">2.4 Naming conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Naming-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Naming-conventions" aria-hidden="true">TOC</a></span></h3>

<p>Names of functions, variables, and struct members must be lowercase, using
underscores (_) to separate words. For example, &lsquo;<samp>avfilter_get_video_buffer</samp>&rsquo;
is an acceptable function name and &lsquo;<samp>AVFilterGetVideo</samp>&rsquo; is not.
</p>
<p>Struct, union, enum, and typedeffed type names must use CamelCase. All structs
and unions should be typedeffed to the same name as the struct/union tag, e.g.
<code>typedef struct AVFoo { ... } AVFoo;</code>. Enums are typically not
typedeffed.
</p>
<p>Enumeration constants and macros must be UPPERCASE, except for macros
masquerading as functions, which should use the function naming convention.
</p>
<p>All identifiers in the libraries should be namespaced as follows:
</p><ul>
<li> No namespacing for identifiers with file and lower scope (e.g. local variables,
static functions), and struct and union members,

</li><li> The <code>ff_</code> prefix must be used for variables and functions visible outside
of file scope, but only used internally within a single library, e.g.
&lsquo;<samp>ff_w64_demuxer</samp>&rsquo;. This prevents name collisions when FFmpeg is statically
linked.

</li><li> For variables and functions visible outside of file scope, used internally
across multiple libraries, use <code>avpriv_</code> as prefix, for example,
&lsquo;<samp>avpriv_report_missing_feature</samp>&rsquo;.

</li><li> All other internal identifiers, like private type or macro names, should be
namespaced only to avoid possible internal conflicts. E.g. <code>H264_NAL_SPS</code>
vs. <code>HEVC_NAL_SPS</code>.

</li><li> Each library has its own prefix for public symbols, in addition to the
commonly used <code>av_</code> (<code>avformat_</code> for libavformat,
<code>avcodec_</code> for libavcodec, <code>swr_</code> for libswresample, etc).
Check the existing code and choose names accordingly.

</li><li> Other public identifiers (struct, union, enum, macro, type names) must use their
library&rsquo;s public prefix (<code>AV</code>, <code>Sws</code>, or <code>Swr</code>).
</li></ul>

<p>Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in <code>_t</code> are reserved by
<a href="http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02">POSIX</a>.
Also avoid names starting with <code>__</code> or <code>_</code> followed by an uppercase
letter as they are reserved by the C standard. Names starting with <code>_</code>
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with <code>_</code> altogether.
</p>
<a name="Miscellaneous-conventions"></a>
<h3 class="section">2.5 Miscellaneous conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Miscellaneous-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Miscellaneous-conventions" aria-hidden="true">TOC</a></span></h3>

<ul>
<li> fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.

</li><li> Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don&rsquo;t make the code easier to understand.
</li></ul>

<span id="Development-Policy"></span><a name="Development-Policy-1"></a>
<h2 class="chapter">3 Development Policy<span class="pull-right"><a class="anchor hidden-xs" href="#Development-Policy-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development-Policy-1" aria-hidden="true">TOC</a></span></h2>

<a name="Patches_002fCommitting"></a>
<h3 class="section">3.1 Patches/Committing<span class="pull-right"><a class="anchor hidden-xs" href="#Patches_002fCommitting" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patches_002fCommitting" aria-hidden="true">TOC</a></span></h3>
<a name="Licenses-for-patches-must-be-compatible-with-FFmpeg_002e"></a>
<p>Contributions should be licensed under the
<a href="http://www.gnu.org/licenses/lgpl-2.1.html">LGPL 2.1</a>,
including an &quot;or any later version&quot; clause, or, if you prefer
a gift-style license, the
<a href="http://opensource.org/licenses/isc-license.txt">ISC</a> or
<a href="http://mit-license.org/">MIT</a> license.
<a href="http://www.gnu.org/licenses/gpl-2.0.html">GPL 2</a> including
an &quot;or any later version&quot; clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
</p>
<a name="You-must-not-commit-code-which-breaks-FFmpeg_0021"></a>
<p>This means unfinished code which is enabled and breaks compilation,
or compiles but does not work/breaks the regression tests. Code which
is unfinished but disabled may be permitted under-circumstances, like
missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
</p>
<a name="Keep-the-main-commit-message-short-with-an-extended-description-below_002e"></a>
<p>The commit message should have a short first line in the form of
a &lsquo;<samp>topic: short description</samp>&rsquo; as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
</p>
<a name="Testing-must-be-adequate-but-not-excessive_002e"></a>
<p>If it works for you, others, and passes FATE then it should be OK to commit
it, provided it fits the other committing criteria. You should not worry about
over-testing things. If your code has problems (portability, triggers
compiler bugs, unusual environment etc) they will be reported and eventually
fixed.
</p>
<a name="Do-not-commit-unrelated-changes-together_002e"></a>
<p>They should be split them into self-contained pieces. Also do not forget
that if part B depends on part A, but A does not depend on B, then A can
and should be committed first and separate from B. Keeping changes well
split into self-contained parts makes reviewing and understanding them on
the commit log mailing list easier. This also helps in case of debugging
later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
</p>
<a name="Ask-before-you-change-the-build-system-_0028configure_002c-etc_0029_002e"></a>
<p>Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
</p>
<a name="Cosmetic-changes-should-be-kept-in-separate-patches_002e"></a>
<p>We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
</p>
<p>NOTE: If you had to put if(){ .. } over a large (&gt; 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
</p>
<a name="Commit-messages-should-always-be-filled-out-properly_002e"></a>
<p>Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as &quot;fixed!&quot; or &quot;Changed it.&quot; are unacceptable.
Recommended format:
</p>
<div class="example">
<pre class="example">area changed: Short 1 line description

details describing what and why and giving references.
</pre></div>

<a name="Credit-the-author-of-the-patch_002e"></a>
<p>Make sure the author of the commit is set correctly. (see git commit &ndash;author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
</p>
<a name="Complex-patches-should-refer-to-discussion-surrounding-them_002e"></a>
<p>When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
</p>
<a name="Always-wait-long-enough-before-pushing-changes"></a>
<p>Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel. If no one answers within a reasonable
time-frame (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
</p>
<a name="Code"></a>
<h3 class="section">3.2 Code<span class="pull-right"><a class="anchor hidden-xs" href="#Code" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code" aria-hidden="true">TOC</a></span></h3>
<a name="API_002fABI-changes-should-be-discussed-before-they-are-made_002e"></a>
<p>Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
</p>
<a name="Remember-to-check-if-you-need-to-bump-versions-for-libav_002a_002e"></a>
<p>Depending on the change, you may need to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
</p>
<a name="Warnings-for-correct-code-may-be-disabled-if-there-is-no-other-option_002e"></a>
<p>Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
</p>
<a name="Check-untrusted-input-properly_002e"></a>
<p>Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
</p>
<a name="Documentation_002fOther"></a>
<h3 class="section">3.3 Documentation/Other<span class="pull-right"><a class="anchor hidden-xs" href="#Documentation_002fOther" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Documentation_002fOther" aria-hidden="true">TOC</a></span></h3>
<a name="Subscribe-to-the-ffmpeg_002ddevel-mailing-list_002e"></a>
<p>It is important to be subscribed to the
<a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel">ffmpeg-devel</a>
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
</p>
<p>However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don&rsquo;t
want to subscribe and discuss the patch, then please do send it to the list
anyway.
</p>
<a name="Subscribe-to-the-ffmpeg_002dcvslog-mailing-list_002e"></a>
<p>Diffs of all commits are sent to the
<a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog">ffmpeg-cvslog</a>
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
</p>
<a name="Keep-the-documentation-up-to-date_002e"></a>
<p>Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
</p>
<a name="Important-discussions-should-be-accessible-to-all_002e"></a>
<p>Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
</p>
<a name="Check-your-entries-in-MAINTAINERS_002e"></a>
<p>Make sure that no parts of the codebase that you maintain are missing from the
<samp>MAINTAINERS</samp> file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help in
finding a new maintainer and also don&rsquo;t forget to update the <samp>MAINTAINERS</samp> file.
</p>
<p>We think our rules are not too hard. If you have comments, contact us.
</p>
<span id="Submitting-patches"></span><a name="Submitting-patches-1"></a>
<h2 class="chapter">4 Submitting patches<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-patches-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-patches-1" aria-hidden="true">TOC</a></span></h2>

<p>First, read the <a href="#Coding-Rules">Coding Rules</a> above if you did not yet, in particular
the rules regarding patch submission.
</p>
<p>When you submit your patch, please use <code>git format-patch</code> or
<code>git send-email</code>. We cannot read other diffs :-).
</p>
<p>Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
</p>
<p>Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
</p>
<p>Run the <a href="#Regression-tests">Regression tests</a> before submitting a patch in order to verify
it does not cause unexpected problems.
</p>
<p>It also helps quite a bit if you tell us what the patch does (for example
&rsquo;replaces lrint by lrintf&rsquo;), and why (for example &rsquo;*BSD isn&rsquo;t C99 compliant
and has no lrint()&rsquo;)
</p>
<p>Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
</p>
<p>Patches should be posted to the
<a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel">ffmpeg-devel</a>
mailing list. Use <code>git send-email</code> when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission. Also ensure the correct mime type is used
(text/x-diff or text/x-patch or at least text/plain) and that only one
patch is inline or attached per mail.
You can check <a href="https://patchwork.ffmpeg.org">https://patchwork.ffmpeg.org</a>, if your patch does not show up, its mime type
likely was wrong.
</p>
<a name="Sending-patches-from-email-clients"></a>
<p>Using <code>git send-email</code> might not be desirable for everyone. The
following trick allows to send patches via email clients in a safe
way. It has been tested with Outlook and Thunderbird (with X-Unsent
extension) and might work with other applications.
</p>
<p>Create your patch like this:
</p>
<pre class="verbatim">git format-patch -s -o &quot;outputfolder&quot; --add-header &quot;X-Unsent: 1&quot; --suffix .eml --to ffmpeg-devel@ffmpeg.org -1 1a2b3c4d
</pre>
<p>Now you&rsquo;ll just need to open the eml file with the email application
and execute &rsquo;Send&rsquo;.
</p>
<a name="Reviews"></a>
<p>Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
</p>
<p>Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
</p>

<a name="New-codecs-or-formats-checklist"></a>
<h2 class="chapter">5 New codecs or formats checklist<span class="pull-right"><a class="anchor hidden-xs" href="#New-codecs-or-formats-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-New-codecs-or-formats-checklist" aria-hidden="true">TOC</a></span></h2>

<ol>
<li> Did you use av_cold for codec initialization and close functions?

</li><li> Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?

</li><li> Did you bump the minor version number (and reset the micro version
number) in <samp>libavcodec/version.h</samp> or <samp>libavformat/version.h</samp>?

</li><li> Did you register it in <samp>allcodecs.c</samp> or <samp>allformats.c</samp>?

</li><li> Did you add the AVCodecID to <samp>avcodec.h</samp>?
When adding new codec IDs, also add an entry to the codec descriptor
list in <samp>libavcodec/codec_desc.c</samp>.

</li><li> If it has a FourCC, did you add it to <samp>libavformat/riff.c</samp>,
even if it is only a decoder?

</li><li> Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you&rsquo;re just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.

</li><li> Did you add an entry to the table of supported formats or codecs in
<samp>doc/general.texi</samp>?

</li><li> Did you add an entry in the Changelog?

</li><li> If it depends on a parser or a library, did you add that dependency in
configure?

</li><li> Did you <code>git add</code> the appropriate files before committing?

</li><li> Did you make sure it compiles standalone, i.e. with
<code>configure --disable-everything --enable-decoder=foo</code>
(or <code>--enable-demuxer</code> or whatever your component is)?
</li></ol>


<a name="Patch-submission-checklist"></a>
<h2 class="chapter">6 Patch submission checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Patch-submission-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patch-submission-checklist" aria-hidden="true">TOC</a></span></h2>

<ol>
<li> Does <code>make fate</code> pass with the patch applied?

</li><li> Was the patch generated with git format-patch or send-email?

</li><li> Did you sign-off your patch? (<code>git commit -s</code>)
See <a href="https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst">Sign your work</a> for the meaning
of <em>sign-off</em>.

</li><li> Did you provide a clear git commit log message?

</li><li> Is the patch against latest FFmpeg git master branch?

</li><li> Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)

</li><li> Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?

</li><li> If the change is to speed critical code, did you benchmark it?

</li><li> If you did any benchmarks, did you provide them in the mail?

</li><li> Have you checked that the patch does not introduce buffer overflows or
other security issues?

</li><li> Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
<a href="http://caca.zoy.org/wiki/zzuf">zzuf</a>. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.

</li><li> Did you test your decoder or demuxer against sample files?
Samples may be obtained at <a href="https://samples.ffmpeg.org">https://samples.ffmpeg.org</a>.

</li><li> Does the patch not mix functional and cosmetic changes?

</li><li> Did you add tabs or trailing whitespace to the code? Both are forbidden.

</li><li> Is the patch attached to the email you send?

</li><li> Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.

</li><li> If the patch fixes a bug, did you provide a verbose analysis of the bug?

</li><li> If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples &gt;100k to mails but rather provide a
URL, you can upload to <a href="https://streams.videolan.org/upload/">https://streams.videolan.org/upload/</a>.

</li><li> Did you provide a verbose summary about what the patch does change?

</li><li> Did you provide a verbose explanation why it changes things like it does?

</li><li> Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?

</li><li> Did you provide an example so we can verify the new feature added by the
patch easily?

</li><li> If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.

</li><li> You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.

</li><li> Lines with similar content should be aligned vertically when doing so
improves readability.

</li><li> Consider adding a regression test for your code.

</li><li> If you added YASM code please check that things still work with &ndash;disable-yasm.

</li><li> Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like <code>av_malloc()</code>
are notoriously left unchecked, which is a serious problem.

</li><li> Test your code with valgrind and or Address Sanitizer to ensure it&rsquo;s free
of leaks, out of array accesses, etc.
</li></ol>

<a name="Patch-review-process"></a>
<h2 class="chapter">7 Patch review process<span class="pull-right"><a class="anchor hidden-xs" href="#Patch-review-process" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patch-review-process" aria-hidden="true">TOC</a></span></h2>

<p>All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
</p>
<p>We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
</p>
<p>If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
</p>
<p>When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
</p>
<p>Everyone is welcome to review patches. Also if you are waiting for your patch
to be reviewed, please consider helping to review other patches, that is a great
way to get everyone&rsquo;s patches reviewed sooner.
</p>
<span id="Regression-tests"></span><a name="Regression-tests-1"></a>
<h2 class="chapter">8 Regression tests<span class="pull-right"><a class="anchor hidden-xs" href="#Regression-tests-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Regression-tests-1" aria-hidden="true">TOC</a></span></h2>

<p>Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
</p>
<p>Running &rsquo;make fate&rsquo; accomplishes this, please see <a href="fate.html">fate.html</a> for details.
</p>
<p>[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
</p>
<a name="Adding-files-to-the-fate_002dsuite-dataset"></a>
<h3 class="section">8.1 Adding files to the fate-suite dataset<span class="pull-right"><a class="anchor hidden-xs" href="#Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">TOC</a></span></h3>

<p>When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
</p>
<a name="Visualizing-Test-Coverage"></a>
<h3 class="section">8.2 Visualizing Test Coverage<span class="pull-right"><a class="anchor hidden-xs" href="#Visualizing-Test-Coverage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Visualizing-Test-Coverage" aria-hidden="true">TOC</a></span></h3>

<p>The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools <code>gcov</code>/<code>lcov</code>. This involves
the following steps:
</p>
<ol>
<li> Configure to compile with instrumentation enabled:
<code>configure --toolchain=gcov</code>.

</li><li> Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.

</li><li> Run <code>make lcov</code> to generate coverage data in HTML format.

</li><li> View <code>lcov/index.html</code> in your preferred HTML viewer.
</li></ol>

<p>You can use the command <code>make lcov-reset</code> to reset the coverage
measurements. You will need to rerun <code>make lcov</code> after running a
new test.
</p>
<a name="Using-Valgrind"></a>
<h3 class="section">8.3 Using Valgrind<span class="pull-right"><a class="anchor hidden-xs" href="#Using-Valgrind" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-Valgrind" aria-hidden="true">TOC</a></span></h3>

<p>The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
<code>--toolchain=valgrind-memcheck</code> or <code>--toolchain=valgrind-massif</code>
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the <strong>memcheck</strong> or the
<strong>massif</strong> tool of the valgrind suite.
</p>
<p>In case you need finer control over how valgrind is invoked, use the
<code>--target-exec='valgrind &lt;your_custom_valgrind_options&gt;</code> option in
your configure line instead.
</p>
<span id="Release-process"></span><a name="Release-process-1"></a>
<h2 class="chapter">9 Release process<span class="pull-right"><a class="anchor hidden-xs" href="#Release-process-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-process-1" aria-hidden="true">TOC</a></span></h2>

<p>FFmpeg maintains a set of <strong>release branches</strong>, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a <strong>release
manager</strong> prepares, tests and publishes tarballs on the
<a href="https://ffmpeg.org">https://ffmpeg.org</a> website.
</p>
<p>There are two kinds of releases:
</p>
<ol>
<li> <strong>Major releases</strong> always include the latest and greatest
features and functionality.

</li><li> <strong>Point releases</strong> are cut from <strong>release</strong> branches,
which are named <code>release/X</code>, with <code>X</code> being the release
version number.
</li></ol>

<p>Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been <strong>compiled</strong> against
previous versions of <strong>the same release series</strong> in any case!
</p>
<p>However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the <strong>ffmpeg-devel</strong> mailing list in time to allow forward planning.
</p>
<span id="Criteria-for-Point-Releases"></span><a name="Criteria-for-Point-Releases-1"></a>
<h3 class="section">9.1 Criteria for Point Releases<span class="pull-right"><a class="anchor hidden-xs" href="#Criteria-for-Point-Releases-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Criteria-for-Point-Releases-1" aria-hidden="true">TOC</a></span></h3>

<p>Changes that match the following criteria are valid candidates for
inclusion into a point release:
</p>
<ol>
<li> Fixes a security issue, preferably identified by a <strong>CVE
number</strong> issued by <a href="http://cve.mitre.org/">http://cve.mitre.org/</a>.

</li><li> Fixes a documented bug in <a href="https://trac.ffmpeg.org">https://trac.ffmpeg.org</a>.

</li><li> Improves the included documentation.

</li><li> Retains both source code and binary compatibility with previous
point releases of the same release branch.
</li></ol>

<p>The order for checking the rules is (1 OR 2 OR 3) AND 4.
</p>

<a name="Release-Checklist"></a>
<h3 class="section">9.2 Release Checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Release-Checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-Checklist" aria-hidden="true">TOC</a></span></h3>

<p>The release process involves the following steps:
</p>
<ol>
<li> Ensure that the <samp>RELEASE</samp> file contains the version number for
the upcoming release.

</li><li> Add the release at <a href="https://trac.ffmpeg.org/admin/ticket/versions">https://trac.ffmpeg.org/admin/ticket/versions</a>.

</li><li> Announce the intent to do a release to the mailing list.

</li><li> Make sure all relevant security fixes have been backported. See
<a href="https://ffmpeg.org/security.html">https://ffmpeg.org/security.html</a>.

</li><li> Ensure that the FATE regression suite still passes in the release
branch on at least <strong>i386</strong> and <strong>amd64</strong>
(cf. <a href="#Regression-tests">Regression tests</a>).

</li><li> Prepare the release tarballs in <code>bz2</code> and <code>gz</code> formats, and
supplementing files that contain <code>gpg</code> signatures

</li><li> Publish the tarballs at <a href="https://ffmpeg.org/releases">https://ffmpeg.org/releases</a>. Create and
push an annotated tag in the form <code>nX</code>, with <code>X</code>
containing the version number.

</li><li> Propose and send a patch to the <strong>ffmpeg-devel</strong> mailing list
with a news entry for the website.

</li><li> Publish the news entry.

</li><li> Send an announcement to the mailing list.
</li></ol>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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FFmpeg FAQ
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FFmpeg FAQ
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-General-Questions" href="#General-Questions">1 General Questions</a>
<ul class="no-bullet">
<li><a id="toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?</a></li>
<li><a id="toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?</a></li>
<li><a id="toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e">1.3 I cannot read this file although this format seems to be supported by ffmpeg.</a></li>
<li><a id="toc-Which-codecs-are-supported-by-Windows_003f" href="#Which-codecs-are-supported-by-Windows_003f">1.4 Which codecs are supported by Windows?</a></li>
</ul></li>
<li><a id="toc-Compilation" href="#Compilation">2 Compilation</a>
<ul class="no-bullet">
<li><a id="toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code></a></li>
<li><a id="toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?</a></li>
<li><a id="toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f">2.3 How do I make <code>pkg-config</code> find my libraries?</a></li>
<li><a id="toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f">2.4 How do I use <code>pkg-config</code> when cross-compiling?</a></li>
</ul></li>
<li><a id="toc-Usage" href="#Usage">3 Usage</a>
<ul class="no-bullet">
<li><a id="toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f">3.1 ffmpeg does not work; what is wrong?</a></li>
<li><a id="toc-How-do-I-encode-single-pictures-into-movies_003f" href="#How-do-I-encode-single-pictures-into-movies_003f">3.2 How do I encode single pictures into movies?</a></li>
<li><a id="toc-How-do-I-encode-movie-to-single-pictures_003f" href="#How-do-I-encode-movie-to-single-pictures_003f">3.3 How do I encode movie to single pictures?</a></li>
<li><a id="toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?</a></li>
<li><a id="toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f">3.5 How can I read from the standard input or write to the standard output?</a></li>
<li><a id="toc-_002df-jpeg-doesn_0027t-work_002e" href="#g_t_002df-jpeg-doesn_0027t-work_002e">3.6 -f jpeg doesn&rsquo;t work.</a></li>
<li><a id="toc-Why-can-I-not-change-the-frame-rate_003f" href="#Why-can-I-not-change-the-frame-rate_003f">3.7 Why can I not change the frame rate?</a></li>
<li><a id="toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f">3.8 How do I encode Xvid or DivX video with ffmpeg?</a></li>
<li><a id="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f">3.9 Which are good parameters for encoding high quality MPEG-4?</a></li>
<li><a id="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?</a></li>
<li><a id="toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?</a></li>
<li><a id="toc-How-can-I-read-DirectShow-files_003f" href="#How-can-I-read-DirectShow-files_003f">3.12 How can I read DirectShow files?</a></li>
<li><a id="toc-How-can-I-join-video-files_003f" href="#How-can-I-join-video-files_003f">3.13 How can I join video files?</a></li>
<li><a id="toc-How-can-I-concatenate-video-files_003f" href="#How-can-I-concatenate-video-files_003f">3.14 How can I concatenate video files?</a>
<ul class="no-bullet">
<li><a id="toc-Concatenating-using-the-concat-filter" href="#Concatenating-using-the-concat-filter">3.14.1 Concatenating using the concat <em>filter</em></a></li>
<li><a id="toc-Concatenating-using-the-concat-demuxer" href="#Concatenating-using-the-concat-demuxer">3.14.2 Concatenating using the concat <em>demuxer</em></a></li>
<li><a id="toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029">3.14.3 Concatenating using the concat <em>protocol</em> (file level)</a></li>
<li><a id="toc-Concatenating-using-raw-audio-and-video" href="#Concatenating-using-raw-audio-and-video">3.14.4 Concatenating using raw audio and video</a></li>
</ul></li>
<li><a id="toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.</a></li>
<li><a id="toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f">3.16 Why does FFmpeg not see the subtitles in my VOB file?</a></li>
<li><a id="toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?</a></li>
<li><a id="toc-I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" href="#I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f">3.18 I have a stretched video, why does scaling does not fix it?</a></li>
<li><a id="toc-How-do-I-run-ffmpeg-as-a-background-task_003f" href="#How-do-I-run-ffmpeg-as-a-background-task_003f">3.19 How do I run ffmpeg as a background task?</a></li>
<li><a id="toc-How-do-I-prevent-ffmpeg-from-suspending-with-a-message-like-suspended-_0028tty-output_0029_003f" href="#How-do-I-prevent-ffmpeg-from-suspending-with-a-message-like-suspended-_0028tty-output_0029_003f">3.20 How do I prevent ffmpeg from suspending with a message like <em>suspended (tty output)</em>?</a></li>
</ul></li>
<li><a id="toc-Development" href="#Development">4 Development</a>
<ul class="no-bullet">
<li><a id="toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?</a></li>
<li><a id="toc-Can-you-support-my-C-compiler-XXX_003f" href="#Can-you-support-my-C-compiler-XXX_003f">4.2 Can you support my C compiler XXX?</a></li>
<li><a id="toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f">4.3 Is Microsoft Visual C++ supported?</a></li>
<li><a id="toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f">4.4 Can you add automake, libtool or autoconf support?</a></li>
<li><a id="toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f">4.5 Why not rewrite FFmpeg in object-oriented C++?</a></li>
<li><a id="toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f">4.6 Why are the ffmpeg programs devoid of debugging symbols?</a></li>
<li><a id="toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f">4.7 I do not like the LGPL, can I contribute code under the GPL instead?</a></li>
<li><a id="toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.</a></li>
<li><a id="toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.</a></li>
<li><a id="toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope</a></li>
<li><a id="toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?</a></li>
<li><a id="toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?</a></li>
<li><a id="toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?</a></li>
<li><a id="toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.</a></li>
<li><a id="toc-Why-is-make-fate-not-running-all-tests_003f" href="#Why-is-make-fate-not-running-all-tests_003f">4.15 Why is <code>make fate</code> not running all tests?</a></li>
<li><a id="toc-Why-is-make-fate-not-finding-the-samples_003f" href="#Why-is-make-fate-not-finding-the-samples_003f">4.16 Why is <code>make fate</code> not finding the samples?</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="General-Questions"></a>
<h2 class="chapter">1 General Questions<span class="pull-right"><a class="anchor hidden-xs" href="#General-Questions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-General-Questions" aria-hidden="true">TOC</a></span></h2>

<a name="Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f"></a>
<h3 class="section">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">TOC</a></span></h3>

<p>Because no one has taken on that task yet. FFmpeg development is
driven by the tasks that are important to the individual developers.
If there is a feature that is important to you, the best way to get
it implemented is to undertake the task yourself or sponsor a developer.
</p>
<a name="FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f"></a>
<h3 class="section">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">TOC</a></span></h3>

<p>No. Windows DLLs are not portable, bloated and often slow.
Moreover FFmpeg strives to support all codecs natively.
A DLL loader is not conducive to that goal.
</p>
<a name="I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e"></a>
<h3 class="section">1.3 I cannot read this file although this format seems to be supported by ffmpeg.<span class="pull-right"><a class="anchor hidden-xs" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">TOC</a></span></h3>

<p>Even if ffmpeg can read the container format, it may not support all its
codecs. Please consult the supported codec list in the ffmpeg
documentation.
</p>
<a name="Which-codecs-are-supported-by-Windows_003f"></a>
<h3 class="section">1.4 Which codecs are supported by Windows?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">TOC</a></span></h3>

<p>Windows does not support standard formats like MPEG very well, unless you
install some additional codecs.
</p>
<p>The following list of video codecs should work on most Windows systems:
</p><dl compact="compact">
<dt><span><samp>msmpeg4v2</samp></span></dt>
<dd><p>.avi/.asf
</p></dd>
<dt><span><samp>msmpeg4</samp></span></dt>
<dd><p>.asf only
</p></dd>
<dt><span><samp>wmv1</samp></span></dt>
<dd><p>.asf only
</p></dd>
<dt><span><samp>wmv2</samp></span></dt>
<dd><p>.asf only
</p></dd>
<dt><span><samp>mpeg4</samp></span></dt>
<dd><p>Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
</p></dd>
<dt><span><samp>mpeg1video</samp></span></dt>
<dd><p>.mpg only
</p></dd>
</dl>
<p>Note, ASF files often have .wmv or .wma extensions in Windows. It should also
be mentioned that Microsoft claims a patent on the ASF format, and may sue
or threaten users who create ASF files with non-Microsoft software. It is
strongly advised to avoid ASF where possible.
</p>
<p>The following list of audio codecs should work on most Windows systems:
</p><dl compact="compact">
<dt><span><samp>adpcm_ima_wav</samp></span></dt>
<dt><span><samp>adpcm_ms</samp></span></dt>
<dt><span><samp>pcm_s16le</samp></span></dt>
<dd><p>always
</p></dd>
<dt><span><samp>libmp3lame</samp></span></dt>
<dd><p>If some MP3 codec like LAME is installed.
</p></dd>
</dl>


<a name="Compilation"></a>
<h2 class="chapter">2 Compilation<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation" aria-hidden="true">TOC</a></span></h2>

<a name="error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027"></a>
<h3 class="section">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code><span class="pull-right"><a class="anchor hidden-xs" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">TOC</a></span></h3>

<p>This is a bug in gcc. Do not report it to us. Instead, please report it to
the gcc developers. Note that we will not add workarounds for gcc bugs.
</p>
<p>Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
<a href="https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203">https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203</a>.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
</p>
<a name="I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f"></a>
<h3 class="section">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">TOC</a></span></h3>

<p>Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
</p>
<p>To build FFmpeg, you need to install the development package. It is usually
called <samp>libfoo-dev</samp> or <samp>libfoo-devel</samp>. You can remove it after the
build is finished, but be sure to keep the main package.
</p>
<a name="How-do-I-make-pkg_002dconfig-find-my-libraries_003f"></a>
<h3 class="section">2.3 How do I make <code>pkg-config</code> find my libraries?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">TOC</a></span></h3>

<p>Somewhere along with your libraries, there is a <samp>.pc</samp> file (or several)
in a <samp>pkgconfig</samp> directory. You need to set environment variables to
point <code>pkg-config</code> to these files.
</p>
<p>If you need to <em>add</em> directories to <code>pkg-config</code>&rsquo;s search list
(typical use case: library installed separately), add it to
<code>$PKG_CONFIG_PATH</code>:
</p>
<div class="example">
<pre class="example">export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
</pre></div>

<p>If you need to <em>replace</em> <code>pkg-config</code>&rsquo;s search list
(typical use case: cross-compiling), set it in
<code>$PKG_CONFIG_LIBDIR</code>:
</p>
<div class="example">
<pre class="example">export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
</pre></div>

<p>If you need to know the library&rsquo;s internal dependencies (typical use: static
linking), add the <code>--static</code> option to <code>pkg-config</code>:
</p>
<div class="example">
<pre class="example">./configure --pkg-config-flags=--static
</pre></div>

<a name="How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f"></a>
<h3 class="section">2.4 How do I use <code>pkg-config</code> when cross-compiling?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">TOC</a></span></h3>

<p>The best way is to install <code>pkg-config</code> in your cross-compilation
environment. It will automatically use the cross-compilation libraries.
</p>
<p>You can also use <code>pkg-config</code> from the host environment by
specifying explicitly <code>--pkg-config=pkg-config</code> to <code>configure</code>.
In that case, you must point <code>pkg-config</code> to the correct directories
using the <code>PKG_CONFIG_LIBDIR</code>, as explained in the previous entry.
</p>
<p>As an intermediate solution, you can place in your cross-compilation
environment a script that calls the host <code>pkg-config</code> with
<code>PKG_CONFIG_LIBDIR</code> set. That script can look like that:
</p>
<div class="example">
<pre class="example">#!/bin/sh
PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
export PKG_CONFIG_LIBDIR
exec /usr/bin/pkg-config &quot;$@&quot;
</pre></div>

<a name="Usage"></a>
<h2 class="chapter">3 Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Usage" aria-hidden="true">TOC</a></span></h2>

<a name="ffmpeg-does-not-work_003b-what-is-wrong_003f"></a>
<h3 class="section">3.1 ffmpeg does not work; what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>

<p>Try a <code>make distclean</code> in the ffmpeg source directory before the build.
If this does not help see
(<a href="https://ffmpeg.org/bugreports.html">https://ffmpeg.org/bugreports.html</a>).
</p>
<a name="How-do-I-encode-single-pictures-into-movies_003f"></a>
<h3 class="section">3.2 How do I encode single pictures into movies?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">TOC</a></span></h3>

<p>First, rename your pictures to follow a numerical sequence.
For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
</pre></div>

<p>Notice that &lsquo;<samp>%d</samp>&rsquo; is replaced by the image number.
</p>
<p><samp>img%03d.jpg</samp> means the sequence <samp>img001.jpg</samp>, <samp>img002.jpg</samp>, etc.
</p>
<p>Use the <samp>-start_number</samp> option to declare a starting number for
the sequence. This is useful if your sequence does not start with
<samp>img001.jpg</samp> but is still in a numerical order. The following
example will start with <samp>img100.jpg</samp>:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
</pre></div>

<p>If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match <code>*jpg</code> to the <samp>/tmp</samp> directory in the sequence of
<samp>img001.jpg</samp>, <samp>img002.jpg</samp> and so on.
</p>
<div class="example">
<pre class="example">x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s &quot;$i&quot; /tmp/img&quot;$counter&quot;.jpg; x=$(($x+1)); done
</pre></div>

<p>If you want to sequence them by oldest modified first, substitute
<code>$(ls -r -t *jpg)</code> in place of <code>*jpg</code>.
</p>
<p>Then run:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
</pre></div>

<p>The same logic is used for any image format that ffmpeg reads.
</p>
<p>You can also use <code>cat</code> to pipe images to ffmpeg:
</p>
<div class="example">
<pre class="example">cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
</pre></div>

<a name="How-do-I-encode-movie-to-single-pictures_003f"></a>
<h3 class="section">3.3 How do I encode movie to single pictures?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">TOC</a></span></h3>

<p>Use:
</p>
<div class="example">
<pre class="example">ffmpeg -i movie.mpg movie%d.jpg
</pre></div>

<p>The <samp>movie.mpg</samp> used as input will be converted to
<samp>movie1.jpg</samp>, <samp>movie2.jpg</samp>, etc...
</p>
<p>Instead of relying on file format self-recognition, you may also use
</p><dl compact="compact">
<dt><span><samp>-c:v ppm</samp></span></dt>
<dt><span><samp>-c:v png</samp></span></dt>
<dt><span><samp>-c:v mjpeg</samp></span></dt>
</dl>
<p>to force the encoding.
</p>
<p>Applying that to the previous example:
</p><div class="example">
<pre class="example">ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
</pre></div>

<p>Beware that there is no &quot;jpeg&quot; codec. Use &quot;mjpeg&quot; instead.
</p>
<a name="Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f"></a>
<h3 class="section">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">TOC</a></span></h3>

<p>For multithreaded MPEG* encoding, the encoded slices must be independent,
otherwise thread n would practically have to wait for n-1 to finish, so it&rsquo;s
quite logical that there is a small reduction of quality. This is not a bug.
</p>
<a name="How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f"></a>
<h3 class="section">3.5 How can I read from the standard input or write to the standard output?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">TOC</a></span></h3>

<p>Use <samp>-</samp> as file name.
</p>
<a name="g_t_002df-jpeg-doesn_0027t-work_002e"></a>
<h3 class="section">3.6 -f jpeg doesn&rsquo;t work.<span class="pull-right"><a class="anchor hidden-xs" href="#_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">TOC</a></span></h3>

<p>Try &rsquo;-f image2 test%d.jpg&rsquo;.
</p>
<a name="Why-can-I-not-change-the-frame-rate_003f"></a>
<h3 class="section">3.7 Why can I not change the frame rate?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">TOC</a></span></h3>

<p>Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v command line option.
</p>
<a name="How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f"></a>
<h3 class="section">3.8 How do I encode Xvid or DivX video with ffmpeg?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">TOC</a></span></h3>

<p>Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
standard (note that there are many other coding formats that use this
same standard). Thus, use &rsquo;-c:v mpeg4&rsquo; to encode in these formats. The
default fourcc stored in an MPEG-4-coded file will be &rsquo;FMP4&rsquo;. If you want
a different fourcc, use the &rsquo;-vtag&rsquo; option. E.g., &rsquo;-vtag xvid&rsquo; will
force the fourcc &rsquo;xvid&rsquo; to be stored as the video fourcc rather than the
default.
</p>
<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f"></a>
<h3 class="section">3.9 Which are good parameters for encoding high quality MPEG-4?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">TOC</a></span></h3>

<p>&rsquo;-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2&rsquo;,
things to try: &rsquo;-bf 2&rsquo;, &rsquo;-mpv_flags qp_rd&rsquo;, &rsquo;-mpv_flags mv0&rsquo;, &rsquo;-mpv_flags skip_rd&rsquo;.
</p>
<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f"></a>
<h3 class="section">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">TOC</a></span></h3>

<p>&rsquo;-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2&rsquo;
but beware the &rsquo;-g 100&rsquo; might cause problems with some decoders.
Things to try: &rsquo;-bf 2&rsquo;, &rsquo;-mpv_flags qp_rd&rsquo;, &rsquo;-mpv_flags mv0&rsquo;, &rsquo;-mpv_flags skip_rd&rsquo;.
</p>
<a name="Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f"></a>
<h3 class="section">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>

<p>You should use &rsquo;-flags +ilme+ildct&rsquo; and maybe &rsquo;-flags +alt&rsquo; for interlaced
material, and try &rsquo;-top 0/1&rsquo; if the result looks really messed-up.
</p>
<a name="How-can-I-read-DirectShow-files_003f"></a>
<h3 class="section">3.12 How can I read DirectShow files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-DirectShow-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-DirectShow-files_003f" aria-hidden="true">TOC</a></span></h3>

<p>If you have built FFmpeg with <code>./configure --enable-avisynth</code>
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
</p>
<p>Just create an &quot;input.avs&quot; text file with this single line ...
</p><div class="example">
<pre class="example">DirectShowSource(&quot;C:\path to your file\yourfile.asf&quot;)
</pre></div>
<p>... and then feed that text file to ffmpeg:
</p><div class="example">
<pre class="example">ffmpeg -i input.avs
</pre></div>

<p>For ANY other help on AviSynth, please visit the
<a href="http://www.avisynth.org/">AviSynth homepage</a>.
</p>
<a name="How-can-I-join-video-files_003f"></a>
<h3 class="section">3.13 How can I join video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-join-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-join-video-files_003f" aria-hidden="true">TOC</a></span></h3>

<p>To &quot;join&quot; video files is quite ambiguous. The following list explains the
different kinds of &quot;joining&quot; and points out how those are addressed in
FFmpeg. To join video files may mean:
</p>
<ul>
<li> To put them one after the other: this is called to <em>concatenate</em> them
(in short: concat) and is addressed
<a href="#How-can-I-concatenate-video-files">in this very faq</a>.

</li><li> To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
<em>multiplex</em> them together (in short: mux), and is done by simply
invoking ffmpeg with several <samp>-i</samp> options.

</li><li> For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
<em>merge</em> them, and can be done using the
<a href="ffmpeg-filters.html#amerge"><code>amerge</code></a> filter.

</li><li> For audio, to play one on top of the other: this is called to <em>mix</em>
them, and can be done by first merging them into a single stream and then
using the <a href="ffmpeg-filters.html#pan"><code>pan</code></a> filter to mix
the channels at will.

</li><li> For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
<a href="ffmpeg-filters.html#overlay"><code>overlay</code></a> video filter.

</li></ul>

<span id="How-can-I-concatenate-video-files"></span><a name="How-can-I-concatenate-video-files_003f"></a>
<h3 class="section">3.14 How can I concatenate video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-concatenate-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-concatenate-video-files_003f" aria-hidden="true">TOC</a></span></h3>

<p>There are several solutions, depending on the exact circumstances.
</p>
<a name="Concatenating-using-the-concat-filter"></a>
<h4 class="subsection">3.14.1 Concatenating using the concat <em>filter</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-filter" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-filter" aria-hidden="true">TOC</a></span></h4>

<p>FFmpeg has a <a href="ffmpeg-filters.html#concat"><code>concat</code></a> filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
</p>
<a name="Concatenating-using-the-concat-demuxer"></a>
<h4 class="subsection">3.14.2 Concatenating using the concat <em>demuxer</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-demuxer" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-demuxer" aria-hidden="true">TOC</a></span></h4>

<p>FFmpeg has a <a href="ffmpeg-formats.html#concat"><code>concat</code></a> demuxer which you can use when you want to avoid a re-encode and
your format doesn&rsquo;t support file level concatenation.
</p>
<a name="Concatenating-using-the-concat-protocol-_0028file-level_0029"></a>
<h4 class="subsection">3.14.3 Concatenating using the concat <em>protocol</em> (file level)<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">TOC</a></span></h4>

<p>FFmpeg has a <a href="ffmpeg-protocols.html#concat"><code>concat</code></a> protocol designed specifically for that, with examples in the
documentation.
</p>
<p>A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
video by merely concatenating the files containing them.
</p>
<p>Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble <code>cat</code> command (or the
equally humble <code>copy</code> under Windows), and finally transcoding back to your
format of choice.
</p>
<div class="example">
<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg &gt; intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
</pre></div>

<p>Additionally, you can use the <code>concat</code> protocol instead of <code>cat</code> or
<code>copy</code> which will avoid creation of a potentially huge intermediate file.
</p>
<div class="example">
<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:&quot;intermediate1.mpg|intermediate2.mpg&quot; -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
</pre></div>

<p>Note that you may need to escape the character &quot;|&quot; which is special for many
shells.
</p>
<p>Another option is usage of named pipes, should your platform support it:
</p>
<div class="example">
<pre class="example">mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg &lt; /dev/null &amp;
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg &lt; /dev/null &amp;
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
</pre></div>

<a name="Concatenating-using-raw-audio-and-video"></a>
<h4 class="subsection">3.14.4 Concatenating using raw audio and video<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-raw-audio-and-video" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-raw-audio-and-video" aria-hidden="true">TOC</a></span></h4>

<p>Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
<code>tail</code> as seen below. Note that when piping through <code>tail</code> you
must use command grouping, <code>{ ;}</code>, to background properly.
</p>
<p>For example, let&rsquo;s say we want to concatenate two FLV files into an
output.flv file:
</p>
<div class="example">
<pre class="example">mkfifo temp1.a
mkfifo temp1.v
mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - &gt; temp1.a &lt; /dev/null &amp;
ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - &gt; temp2.a &lt; /dev/null &amp;
ffmpeg -i input1.flv -an -f yuv4mpegpipe - &gt; temp1.v &lt; /dev/null &amp;
{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - &lt; /dev/null | tail -n +2 &gt; temp2.v ; } &amp;
cat temp1.a temp2.a &gt; all.a &amp;
cat temp1.v temp2.v &gt; all.v &amp;
ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
</pre></div>

<a name="Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e"></a>
<h3 class="section">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.<span class="pull-right"><a class="anchor hidden-xs" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">TOC</a></span></h3>

<p>Use <samp>-dumpgraph -</samp> to find out exactly where the channel layout is
lost.
</p>
<p>Most likely, it is through <code>auto-inserted aresample</code>. Try to understand
why the converting filter was needed at that place.
</p>
<p>Just before the output is a likely place, as <samp>-f lavfi</samp> currently
only support packed S16.
</p>
<p>Then insert the correct <code>aformat</code> explicitly in the filtergraph,
specifying the exact format.
</p>
<div class="example">
<pre class="example">aformat=sample_fmts=s16:channel_layouts=stereo
</pre></div>

<a name="Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f"></a>
<h3 class="section">3.16 Why does FFmpeg not see the subtitles in my VOB file?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">TOC</a></span></h3>

<p>VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initially detected.
</p>
<p>Some applications, including the <code>ffmpeg</code> command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.
</p>
<p>The size of the initial scan is controlled by two options: <code>probesize</code>
(default ~5 Mo) and <code>analyzeduration</code> (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
</p>
<a name="Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f"></a>
<h3 class="section">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">TOC</a></span></h3>

<p>The <samp>-sameq</samp> option meant &quot;same quantizer&quot;, and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
&quot;same quality&quot; and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
</p>
<p>Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
<samp>-qscale</samp> and <samp>-qmax</samp>, but you should peruse the documentation
of the encoder you chose.
</p>
<a name="I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f"></a>
<h3 class="section">3.18 I have a stretched video, why does scaling does not fix it?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" aria-hidden="true">TOC</a></span></h3>

<p>A lot of video codecs and formats can store the <em>aspect ratio</em> of the
video: this is the ratio between the width and the height of either the full
image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
SAR.
</p>
<p>Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
of video standards, especially from the analogic-numeric transition era, use
non-square pixels.
</p>
<p>Most processing filters in FFmpeg handle the aspect ratio to avoid
stretching the image: cropping adjusts the DAR to keep the SAR constant,
scaling adjusts the SAR to keep the DAR constant.
</p>
<p>If you want to stretch, or “unstretch”, the image, you need to override the
information with the
<a href="ffmpeg-filters.html#setdar_002c-setsar"><code>setdar or setsar filters</code></a>.
</p>
<p>Do not forget to examine carefully the original video to check whether the
stretching comes from the image or from the aspect ratio information.
</p>
<p>For example, to fix a badly encoded EGA capture, use the following commands,
either the first one to upscale to square pixels or the second one to set
the correct aspect ratio or the third one to avoid transcoding (may not work
depending on the format / codec / player / phase of the moon):
</p>
<div class="example">
<pre class="example">ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
</pre></div>

<span id="background-task"></span><a name="How-do-I-run-ffmpeg-as-a-background-task_003f"></a>
<h3 class="section">3.19 How do I run ffmpeg as a background task?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-run-ffmpeg-as-a-background-task_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-run-ffmpeg-as-a-background-task_003f" aria-hidden="true">TOC</a></span></h3>

<p>ffmpeg normally checks the console input, for entries like &quot;q&quot; to stop
and &quot;?&quot; to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
</p>
<p>To prevent those input checks, allowing ffmpeg to run as a background task,
use the <a href="ffmpeg.html#stdin-option"><code>-nostdin</code> option</a>
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
</p>
<p>As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to <code>/dev/null</code> (on Linux and macOS)
or <code>NUL</code> (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
</p>
<p>For example:
</p>
<div class="example">
<pre class="example">ffmpeg -nostdin -i INPUT OUTPUT
</pre></div>

<p>or (on Linux, macOS, and other UNIX-like shells):
</p>
<div class="example">
<pre class="example">ffmpeg -i INPUT OUTPUT &lt;/dev/null
</pre></div>

<p>or (on Windows):
</p>
<div class="example">
<pre class="example">ffmpeg -i INPUT OUTPUT &lt;NUL
</pre></div>

<a name="How-do-I-prevent-ffmpeg-from-suspending-with-a-message-like-suspended-_0028tty-output_0029_003f"></a>
<h3 class="section">3.20 How do I prevent ffmpeg from suspending with a message like <em>suspended (tty output)</em>?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-prevent-ffmpeg-from-suspending-with-a-message-like-suspended-_0028tty-output_0029_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-prevent-ffmpeg-from-suspending-with-a-message-like-suspended-_0028tty-output_0029_003f" aria-hidden="true">TOC</a></span></h3>

<p>If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like <em>suspended (tty output)</em>. The question is how
to prevent the process from being suspended.
</p>
<p>For example:
</p>
<div class="example">
<pre class="example">% ffmpeg -i INPUT OUTPUT &amp;&gt; ~/tmp/log.txt &amp;
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &amp;&gt;
</pre></div>

<p>The message &quot;tty output&quot; notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
</p>
<p>The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
<a href="ffmpeg.html#stdin-option"><code>-nostdin</code> option</a>,
or redirect standard input with <code>&lt; /dev/null</code>.
See FAQ
<a href="#background-task"><em>How do I run ffmpeg as a background task?</em></a>
for details.
</p>
<a name="Development"></a>
<h2 class="chapter">4 Development<span class="pull-right"><a class="anchor hidden-xs" href="#Development" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development" aria-hidden="true">TOC</a></span></h2>

<a name="Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f"></a>
<h3 class="section">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">TOC</a></span></h3>

<p>Yes. Check the <samp>doc/examples</samp> directory in the source
repository, also available online at:
<a href="https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples">https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples</a>.
</p>
<p>Examples are also installed by default, usually in
<code>$PREFIX/share/ffmpeg/examples</code>.
</p>
<p>Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
examine the source code for one of the many open source projects that
already incorporate FFmpeg at (<a href="projects.html">projects.html</a>).
</p>
<a name="Can-you-support-my-C-compiler-XXX_003f"></a>
<h3 class="section">4.2 Can you support my C compiler XXX?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">TOC</a></span></h3>

<p>It depends. If your compiler is C99-compliant, then patches to support
it are likely to be welcome if they do not pollute the source code
with <code>#ifdef</code>s related to the compiler.
</p>
<a name="Is-Microsoft-Visual-C_002b_002b-supported_003f"></a>
<h3 class="section">4.3 Is Microsoft Visual C++ supported?<span class="pull-right"><a class="anchor hidden-xs" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">TOC</a></span></h3>

<p>Yes. Please see the <a href="platform.html">Microsoft Visual C++</a>
section in the FFmpeg documentation.
</p>
<a name="Can-you-add-automake_002c-libtool-or-autoconf-support_003f"></a>
<h3 class="section">4.4 Can you add automake, libtool or autoconf support?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">TOC</a></span></h3>

<p>No. These tools are too bloated and they complicate the build.
</p>
<a name="Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f"></a>
<h3 class="section">4.5 Why not rewrite FFmpeg in object-oriented C++?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read <a href="https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15">&quot;Programming Religion&quot;</a>.
</p>
<a name="Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f"></a>
<h3 class="section">4.6 Why are the ffmpeg programs devoid of debugging symbols?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">TOC</a></span></h3>

<p>The build process creates <code>ffmpeg_g</code>, <code>ffplay_g</code>, etc. which
contain full debug information. Those binaries are stripped to create
<code>ffmpeg</code>, <code>ffplay</code>, etc. If you need the debug information, use
the *_g versions.
</p>
<a name="I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f"></a>
<h3 class="section">4.7 I do not like the LGPL, can I contribute code under the GPL instead?<span class="pull-right"><a class="anchor hidden-xs" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">TOC</a></span></h3>

<p>Yes, as long as the code is optional and can easily and cleanly be placed
under #if CONFIG_GPL without breaking anything. So, for example, a new codec
or filter would be OK under GPL while a bug fix to LGPL code would not.
</p>
<a name="I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e"></a>
<h3 class="section">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: <code>-lavdevice</code> must come before
<code>-lavformat</code>, <code>-lavutil</code> must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.
</p>
<p>An easy way to get the full list of required libraries in dependency order
is to use <code>pkg-config</code>.
</p>
<div class="example">
<pre class="example">c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
</pre></div>

<p>See <samp>doc/example/Makefile</samp> and <samp>doc/example/pc-uninstalled</samp> for
more details.
</p>
<a name="I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e"></a>
<h3 class="section">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg is a pure C project, so to use the libraries within your C++ application
you need to explicitly state that you are using a C library. You can do this by
encompassing your FFmpeg includes using <code>extern &quot;C&quot;</code>.
</p>
<p>See <a href="http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3">http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3</a>
</p>
<a name="I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope"></a>
<h3 class="section">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg is a pure C project using C99 math features, in order to enable C++
to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
</p>
<a name="I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f"></a>
<h3 class="section">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">TOC</a></span></h3>

<p>You have to create a custom AVIOContext using <code>avio_alloc_context</code>,
see <samp>libavformat/aviobuf.c</samp> in FFmpeg and <samp>libmpdemux/demux_lavf.c</samp> in MPlayer or MPlayer2 sources.
</p>
<a name="Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f"></a>
<h3 class="section">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?<span class="pull-right"><a class="anchor hidden-xs" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">TOC</a></span></h3>

<p>see <a href="https://www.ffmpeg.org/~michael/">https://www.ffmpeg.org/~michael/</a>
</p>
<a name="How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f"></a>
<h3 class="section">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">TOC</a></span></h3>

<p>Even if peculiar since it is network oriented, RTP is a container like any
other. You have to <em>demux</em> RTP before feeding the payload to libavcodec.
In this specific case please look at RFC 4629 to see how it should be done.
</p>
<a name="AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e"></a>
<h3 class="section">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.<span class="pull-right"><a class="anchor hidden-xs" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">TOC</a></span></h3>

<p><code>r_frame_rate</code> is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then <code>r_frame_rate</code>
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see <code>AVStream.avg_frame_rate</code>.
</p>
<a name="Why-is-make-fate-not-running-all-tests_003f"></a>
<h3 class="section">4.15 Why is <code>make fate</code> not running all tests?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">TOC</a></span></h3>

<p>Make sure you have the fate-suite samples and the <code>SAMPLES</code> Make variable
or <code>FATE_SAMPLES</code> environment variable or the <code>--samples</code>
<code>configure</code> option is set to the right path.
</p>
<a name="Why-is-make-fate-not-finding-the-samples_003f"></a>
<h3 class="section">4.16 Why is <code>make fate</code> not finding the samples?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">TOC</a></span></h3>

<p>Do you happen to have a <code>~</code> character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace <code>~</code> by the full path.
</p>
<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Introduction" href="#Introduction">1 Introduction</a></li>
<li><a id="toc-Using-FATE-from-your-FFmpeg-source-directory" href="#Using-FATE-from-your-FFmpeg-source-directory">2 Using FATE from your FFmpeg source directory</a></li>
<li><a id="toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server">3 Submitting the results to the FFmpeg result aggregation server</a></li>
<li><a id="toc-Uploading-new-samples-to-the-fate-suite" href="#Uploading-new-samples-to-the-fate-suite">4 Uploading new samples to the fate suite</a></li>
<li><a id="toc-FATE-makefile-targets-and-variables" href="#FATE-makefile-targets-and-variables">5 FATE makefile targets and variables</a>
<ul class="no-bullet">
<li><a id="toc-Makefile-targets" href="#Makefile-targets">5.1 Makefile targets</a></li>
<li><a id="toc-Makefile-variables" href="#Makefile-variables">5.2 Makefile variables</a></li>
<li><a id="toc-Examples" href="#Examples">5.3 Examples</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="Introduction"></a>
<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>

<p>FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
</p>
<p>The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg&rsquo;s
FATE server.
</p>
<p>In any way you can have a look at the publicly viewable FATE results
by visiting this website:
</p>
<p><a href="http://fate.ffmpeg.org/">http://fate.ffmpeg.org/</a>
</p>
<p>This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
</p>
<p>The second part of this document describes how you can run FATE to
submit your results to FFmpeg&rsquo;s FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
</p>
<p>In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
</p>

<a name="Using-FATE-from-your-FFmpeg-source-directory"></a>
<h2 class="chapter">2 Using FATE from your FFmpeg source directory<span class="pull-right"><a class="anchor hidden-xs" href="#Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">TOC</a></span></h2>

<p>If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
</p>
<div class="example">
<pre class="example">make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
</pre></div>

<p>The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
<samp>--samples=&lt;path to the samples directory&gt;</samp>. Afterwards you can
invoke the makefile targets without setting the <var>SAMPLES</var> makefile
variable. This is illustrated by the following commands:
</p>
<div class="example">
<pre class="example">./configure --samples=fate-suite/
make fate-rsync
make fate
</pre></div>

<p>Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
</p>
<div class="example">
<pre class="example">FATE_SAMPLES=fate-suite/ make fate
</pre></div>

<div class="info">
<p>Do not put a &rsquo;~&rsquo; character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
</p></div>
<p>To get the complete list of tests, run the command:
</p><div class="example">
<pre class="example">make fate-list
</pre></div>

<p>You can specify a subset of tests to run by specifying the
corresponding elements from the list with the <code>fate-</code> prefix,
e.g. as in:
</p><div class="example">
<pre class="example">make fate-ffprobe_compact fate-ffprobe_xml
</pre></div>

<p>This makes it easier to run a few tests in case of failure without
running the complete test suite.
</p>
<p>To use a custom wrapper to run the test, pass <samp>--target-exec</samp> to
<code>configure</code> or set the <var>TARGET_EXEC</var> Make variable.
</p>

<a name="Submitting-the-results-to-the-FFmpeg-result-aggregation-server"></a>
<h2 class="chapter">3 Submitting the results to the FFmpeg result aggregation server<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">TOC</a></span></h2>

<p>To submit your results to the server you should run fate through the
shell script <samp>tests/fate.sh</samp> from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
</p>
<div class="example">
<pre class="example">tests/fate.sh /path/to/fate_config
</pre></div>

<p>A configuration file template with comments describing the individual
configuration variables can be found at <samp>doc/fate_config.sh.template</samp>.
</p>
<p>The mentioned configuration template is also available here:
</p><pre class="verbatim">slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
#branch=release/2.6 # the branch to test
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv=&quot;ssh -T fate@fate.ffmpeg.org&quot; # command to submit report
comment= # optional description
build_only= # set to &quot;yes&quot; for a compile-only instance that skips tests
ignore_tests=

# the following are optional and map to configure options
arch=
cpu=
cross_prefix=
as=
cc=
ld=
target_os=
sysroot=
target_exec=
target_path=
target_samples=
extra_cflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above

#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#makeopts_fate= # extra options passed to 'make' when running tests,
# defaulting to makeopts above if this is not set
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'
</pre>
<p>Create a configuration that suits your needs, based on the configuration
template. The <code>slot</code> configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern &lsquo;<samp><var>arch</var>-<var>os</var>-<var>compiler</var>-<var>compiler version</var></samp>&rsquo;. The
configuration file itself will be sourced in a shell script, therefore all
shell features may be used. This enables you to setup the environment as you
need it for your build.
</p>
<p>For your first test runs the <code>fate_recv</code> variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
</p>
<ul>
<li> configure.log
</li><li> compile.log
</li><li> test.log
</li><li> report
</li><li> version
</li></ul>

<p>When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address <a href="mailto:fate-admin@ffmpeg.org">fate-admin@ffmpeg.org</a>.
</p>
<p>Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server&rsquo;s fingerprint is:
</p>
<dl compact="compact">
<dt><span>&lsquo;<samp>RSA</samp>&rsquo;</span></dt>
<dd><p>d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
</p></dd>
<dt><span>&lsquo;<samp>ECDSA</samp>&rsquo;</span></dt>
<dd><p>76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
</p></dd>
</dl>

<p>If you have problems connecting to the FATE server, it may help to try out
the <code>ssh</code> command with one or more <samp>-v</samp> options. You should
get detailed output concerning your SSH configuration and the authentication
process.
</p>
<p>The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
</p>
<a name="Uploading-new-samples-to-the-fate-suite"></a>
<h2 class="chapter">4 Uploading new samples to the fate suite<span class="pull-right"><a class="anchor hidden-xs" href="#Uploading-new-samples-to-the-fate-suite" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Uploading-new-samples-to-the-fate-suite" aria-hidden="true">TOC</a></span></h2>

<p>If you need a sample uploaded send a mail to samples-request.
</p>
<p>This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
Also all needed samples for a commit should be uploaded, ideally 24
hours, before the push.
If you need an account for frequently uploading samples or you wish to help
others by doing that send a mail to ffmpeg-devel.
</p>
<div class="example">
<pre class="example">#First update your local samples copy:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite

#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite

#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
</pre></div>


<a name="FATE-makefile-targets-and-variables"></a>
<h2 class="chapter">5 FATE makefile targets and variables<span class="pull-right"><a class="anchor hidden-xs" href="#FATE-makefile-targets-and-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FATE-makefile-targets-and-variables" aria-hidden="true">TOC</a></span></h2>

<a name="Makefile-targets"></a>
<h3 class="section">5.1 Makefile targets<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-targets" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-targets" aria-hidden="true">TOC</a></span></h3>

<dl compact="compact">
<dt><span><samp>fate-rsync</samp></span></dt>
<dd><p>Download/synchronize sample files to the configured samples directory.
</p>
</dd>
<dt><span><samp>fate-list</samp></span></dt>
<dd><p>Will list all fate/regression test targets.
</p>
</dd>
<dt><span><samp>fate</samp></span></dt>
<dd><p>Run the FATE test suite (requires the fate-suite dataset).
</p></dd>
</dl>

<a name="Makefile-variables"></a>
<h3 class="section">5.2 Makefile variables<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-variables" aria-hidden="true">TOC</a></span></h3>

<dl compact="compact">
<dt><span><code>V</code></span></dt>
<dd><p>Verbosity level, can be set to 0, 1 or 2.
</p><ul>
<li> 0: show just the test arguments
</li><li> 1: show just the command used in the test
</li><li> 2: show everything
</li></ul>

</dd>
<dt><span><code>SAMPLES</code></span></dt>
<dd><p>Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
</p>
</dd>
<dt><span><code>THREADS</code></span></dt>
<dd><p>Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
</p>
</dd>
<dt><span><code>THREAD_TYPE</code></span></dt>
<dd><p>Specify which threading strategy test, either &lsquo;<samp>slice</samp>&rsquo; or &lsquo;<samp>frame</samp>&rsquo;,
by default &lsquo;<samp>slice+frame</samp>&rsquo;
</p>
</dd>
<dt><span><code>CPUFLAGS</code></span></dt>
<dd><p>Specify CPU flags.
</p>
</dd>
<dt><span><code>TARGET_EXEC</code></span></dt>
<dd><p>Specify or override the wrapper used to run the tests.
The <code>TARGET_EXEC</code> option provides a way to run FATE wrapped in
<code>valgrind</code>, <code>qemu-user</code> or <code>wine</code> or on remote targets
through <code>ssh</code>.
</p>
</dd>
<dt><span><code>GEN</code></span></dt>
<dd><p>Set to &lsquo;<samp>1</samp>&rsquo; to generate the missing or mismatched references.
</p>
</dd>
<dt><span><code>HWACCEL</code></span></dt>
<dd><p>Specify which hardware acceleration to use while running regression tests,
by default &lsquo;<samp>none</samp>&rsquo; is used.
</p>
</dd>
<dt><span><code>KEEP</code></span></dt>
<dd><p>Set to &lsquo;<samp>1</samp>&rsquo; to keep temp files generated by fate test(s) when test is successful.
Default is &lsquo;<samp>0</samp>&rsquo;, which removes these files. Files are always kept when a test
fails.
</p>
</dd>
</dl>

<a name="Examples"></a>
<h3 class="section">5.3 Examples<span class="pull-right"><a class="anchor hidden-xs" href="#Examples" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Examples" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
</pre></div>
<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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FFmpeg Resampler Documentation
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FFmpeg Resampler Documentation
</h1>
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li>
<li><a id="toc-See-Also" href="#See-Also">3 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">4 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg resampler provides a high-level interface to the
libswresample library audio resampling utilities. In particular it
allows one to perform audio resampling, audio channel layout rematrixing,
and convert audio format and packing layout.
</p>

<a name="Resampler-Options"></a>
<h2 class="chapter">2 Resampler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Resampler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Resampler-Options" aria-hidden="true">TOC</a></span></h2>

<p>The audio resampler supports the following named options.
</p>
<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
FFmpeg tools, <var>option</var>=<var>value</var> for the aresample filter,
by setting the value explicitly in the
<code>SwrContext</code> options or using the <samp>libavutil/opt.h</samp> API for
programmatic use.
</p>
<dl compact="compact">
<dt><span><samp>uchl, used_chlayout</samp></span></dt>
<dd><p>Set used input channel layout. Default is unset. This option is
only used for special remapping.
</p>
</dd>
<dt><span><samp>isr, in_sample_rate</samp></span></dt>
<dd><p>Set the input sample rate. Default value is 0.
</p>
</dd>
<dt><span><samp>osr, out_sample_rate</samp></span></dt>
<dd><p>Set the output sample rate. Default value is 0.
</p>
</dd>
<dt><span><samp>isf, in_sample_fmt</samp></span></dt>
<dd><p>Specify the input sample format. It is set by default to <code>none</code>.
</p>
</dd>
<dt><span><samp>osf, out_sample_fmt</samp></span></dt>
<dd><p>Specify the output sample format. It is set by default to <code>none</code>.
</p>
</dd>
<dt><span><samp>tsf, internal_sample_fmt</samp></span></dt>
<dd><p>Set the internal sample format. Default value is <code>none</code>.
This will automatically be chosen when it is not explicitly set.
</p>
</dd>
<dt><span><samp>ichl, in_chlayout</samp></span></dt>
<dt><span><samp>ochl, out_chlayout</samp></span></dt>
<dd><p>Set the input/output channel layout.
</p>
<p>See <a data-manual="ffmpeg-utils" href="ffmpeg-utils.html#channel-layout-syntax">(ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual</a>
for the required syntax.
</p>
</dd>
<dt><span><samp>clev, center_mix_level</samp></span></dt>
<dd><p>Set the center mix level. It is a value expressed in deciBel, and must be
in the interval [-32,32].
</p>
</dd>
<dt><span><samp>slev, surround_mix_level</samp></span></dt>
<dd><p>Set the surround mix level. It is a value expressed in deciBel, and must
be in the interval [-32,32].
</p>
</dd>
<dt><span><samp>lfe_mix_level</samp></span></dt>
<dd><p>Set LFE mix into non LFE level. It is used when there is a LFE input but no
LFE output. It is a value expressed in deciBel, and must
be in the interval [-32,32].
</p>
</dd>
<dt><span><samp>rmvol, rematrix_volume</samp></span></dt>
<dd><p>Set rematrix volume. Default value is 1.0.
</p>
</dd>
<dt><span><samp>rematrix_maxval</samp></span></dt>
<dd><p>Set maximum output value for rematrixing.
This can be used to prevent clipping vs. preventing volume reduction.
A value of 1.0 prevents clipping.
</p>
</dd>
<dt><span><samp>flags, swr_flags</samp></span></dt>
<dd><p>Set flags used by the converter. Default value is 0.
</p>
<p>It supports the following individual flags:
</p><dl compact="compact">
<dt><span><samp>res</samp></span></dt>
<dd><p>force resampling, this flag forces resampling to be used even when the
input and output sample rates match.
</p></dd>
</dl>

</dd>
<dt><span><samp>dither_scale</samp></span></dt>
<dd><p>Set the dither scale. Default value is 1.
</p>
</dd>
<dt><span><samp>dither_method</samp></span></dt>
<dd><p>Set dither method. Default value is 0.
</p>
<p>Supported values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>rectangular</samp>&rsquo;</span></dt>
<dd><p>select rectangular dither
</p></dd>
<dt><span>&lsquo;<samp>triangular</samp>&rsquo;</span></dt>
<dd><p>select triangular dither
</p></dd>
<dt><span>&lsquo;<samp>triangular_hp</samp>&rsquo;</span></dt>
<dd><p>select triangular dither with high pass
</p></dd>
<dt><span>&lsquo;<samp>lipshitz</samp>&rsquo;</span></dt>
<dd><p>select Lipshitz noise shaping dither.
</p></dd>
<dt><span>&lsquo;<samp>shibata</samp>&rsquo;</span></dt>
<dd><p>select Shibata noise shaping dither.
</p></dd>
<dt><span>&lsquo;<samp>low_shibata</samp>&rsquo;</span></dt>
<dd><p>select low Shibata noise shaping dither.
</p></dd>
<dt><span>&lsquo;<samp>high_shibata</samp>&rsquo;</span></dt>
<dd><p>select high Shibata noise shaping dither.
</p></dd>
<dt><span>&lsquo;<samp>f_weighted</samp>&rsquo;</span></dt>
<dd><p>select f-weighted noise shaping dither
</p></dd>
<dt><span>&lsquo;<samp>modified_e_weighted</samp>&rsquo;</span></dt>
<dd><p>select modified-e-weighted noise shaping dither
</p></dd>
<dt><span>&lsquo;<samp>improved_e_weighted</samp>&rsquo;</span></dt>
<dd><p>select improved-e-weighted noise shaping dither
</p>
</dd>
</dl>

</dd>
<dt><span><samp>resampler</samp></span></dt>
<dd><p>Set resampling engine. Default value is swr.
</p>
<p>Supported values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>swr</samp>&rsquo;</span></dt>
<dd><p>select the native SW Resampler; filter options precision and cheby are not
applicable in this case.
</p></dd>
<dt><span>&lsquo;<samp>soxr</samp>&rsquo;</span></dt>
<dd><p>select the SoX Resampler (where available); compensation, and filter options
filter_size, phase_shift, exact_rational, filter_type &amp; kaiser_beta, are not
applicable in this case.
</p></dd>
</dl>

</dd>
<dt><span><samp>filter_size</samp></span></dt>
<dd><p>For swr only, set resampling filter size, default value is 32.
</p>
</dd>
<dt><span><samp>phase_shift</samp></span></dt>
<dd><p>For swr only, set resampling phase shift, default value is 10, and must be in
the interval [0,30].
</p>
</dd>
<dt><span><samp>linear_interp</samp></span></dt>
<dd><p>Use linear interpolation when enabled (the default). Disable it if you want
to preserve speed instead of quality when exact_rational fails.
</p>
</dd>
<dt><span><samp>exact_rational</samp></span></dt>
<dd><p>For swr only, when enabled, try to use exact phase_count based on input and
output sample rate. However, if it is larger than <code>1 &lt;&lt; phase_shift</code>,
the phase_count will be <code>1 &lt;&lt; phase_shift</code> as fallback. Default is enabled.
</p>
</dd>
<dt><span><samp>cutoff</samp></span></dt>
<dd><p>Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
</p>
</dd>
<dt><span><samp>precision</samp></span></dt>
<dd><p>For soxr only, the precision in bits to which the resampled signal will be
calculated. The default value of 20 (which, with suitable dithering, is
appropriate for a destination bit-depth of 16) gives SoX&rsquo;s &rsquo;High Quality&rsquo;; a
value of 28 gives SoX&rsquo;s &rsquo;Very High Quality&rsquo;.
</p>
</dd>
<dt><span><samp>cheby</samp></span></dt>
<dd><p>For soxr only, selects passband rolloff none (Chebyshev) &amp; higher-precision
approximation for &rsquo;irrational&rsquo; ratios. Default value is 0.
</p>
</dd>
<dt><span><samp>async</samp></span></dt>
<dd><p>For swr only, simple 1 parameter audio sync to timestamps using stretching,
squeezing, filling and trimming. Setting this to 1 will enable filling and
trimming, larger values represent the maximum amount in samples that the data
may be stretched or squeezed for each second.
Default value is 0, thus no compensation is applied to make the samples match
the audio timestamps.
</p>
</dd>
<dt><span><samp>first_pts</samp></span></dt>
<dd><p>For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
This allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame&rsquo;s expected pts, so no padding or
trimming is done. For example, this could be set to 0 to pad the beginning with
silence if an audio stream starts after the video stream or to trim any samples
with a negative pts due to encoder delay.
</p>
</dd>
<dt><span><samp>min_comp</samp></span></dt>
<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger stretching/squeezing/filling or trimming of the
data to make it match the timestamps. The default is that
stretching/squeezing/filling and trimming is disabled
(<samp>min_comp</samp> = <code>FLT_MAX</code>).
</p>
</dd>
<dt><span><samp>min_hard_comp</samp></span></dt>
<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger adding/dropping samples to make it match the
timestamps. This option effectively is a threshold to select between
hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
all compensation is by default disabled through <samp>min_comp</samp>.
The default is 0.1.
</p>
</dd>
<dt><span><samp>comp_duration</samp></span></dt>
<dd><p>For swr only, set duration (in seconds) over which data is stretched/squeezed
to make it match the timestamps. Must be a non-negative double float value,
default value is 1.0.
</p>
</dd>
<dt><span><samp>max_soft_comp</samp></span></dt>
<dd><p>For swr only, set maximum factor by which data is stretched/squeezed to make it
match the timestamps. Must be a non-negative double float value, default value
is 0.
</p>
</dd>
<dt><span><samp>matrix_encoding</samp></span></dt>
<dd><p>Select matrixed stereo encoding.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>none</samp>&rsquo;</span></dt>
<dd><p>select none
</p></dd>
<dt><span>&lsquo;<samp>dolby</samp>&rsquo;</span></dt>
<dd><p>select Dolby
</p></dd>
<dt><span>&lsquo;<samp>dplii</samp>&rsquo;</span></dt>
<dd><p>select Dolby Pro Logic II
</p></dd>
</dl>

<p>Default value is <code>none</code>.
</p>
</dd>
<dt><span><samp>filter_type</samp></span></dt>
<dd><p>For swr only, select resampling filter type. This only affects resampling
operations.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>cubic</samp>&rsquo;</span></dt>
<dd><p>select cubic
</p></dd>
<dt><span>&lsquo;<samp>blackman_nuttall</samp>&rsquo;</span></dt>
<dd><p>select Blackman Nuttall windowed sinc
</p></dd>
<dt><span>&lsquo;<samp>kaiser</samp>&rsquo;</span></dt>
<dd><p>select Kaiser windowed sinc
</p></dd>
</dl>

</dd>
<dt><span><samp>kaiser_beta</samp></span></dt>
<dd><p>For swr only, set Kaiser window beta value. Must be a double float value in the
interval [2,16], default value is 9.
</p>
</dd>
<dt><span><samp>output_sample_bits</samp></span></dt>
<dd><p>For swr only, set number of used output sample bits for dithering. Must be an integer in the
interval [0,64], default value is 0, which means it&rsquo;s not used.
</p>
</dd>
</dl>


<a name="See-Also"></a>
<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="libswresample.html">libswresample</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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FFmpeg Scaler Documentation
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FFmpeg Scaler Documentation
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-Scaler-Options" href="#Scaler-Options">2 Scaler Options</a></li>
<li><a id="toc-See-Also" href="#See-Also">3 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">4 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg rescaler provides a high-level interface to the libswscale
library image conversion utilities. In particular it allows one to perform
image rescaling and pixel format conversion.
</p>

<span id="scaler_005foptions"></span><a name="Scaler-Options"></a>
<h2 class="chapter">2 Scaler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Scaler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Scaler-Options" aria-hidden="true">TOC</a></span></h2>

<p>The video scaler supports the following named options.
</p>
<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
FFmpeg tools, with a few API-only exceptions noted below.
For programmatic use, they can be set explicitly in the
<code>SwsContext</code> options or through the <samp>libavutil/opt.h</samp> API.
</p>
<dl compact="compact">
<dd>
<span id="sws_005fflags"></span></dd>
<dt><span><samp>sws_flags</samp></span></dt>
<dd><p>Set the scaler flags. This is also used to set the scaling
algorithm. Only a single algorithm should be selected. Default
value is &lsquo;<samp>bicubic</samp>&rsquo;.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>fast_bilinear</samp>&rsquo;</span></dt>
<dd><p>Select fast bilinear scaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>bilinear</samp>&rsquo;</span></dt>
<dd><p>Select bilinear scaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>bicubic</samp>&rsquo;</span></dt>
<dd><p>Select bicubic scaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>experimental</samp>&rsquo;</span></dt>
<dd><p>Select experimental scaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>neighbor</samp>&rsquo;</span></dt>
<dd><p>Select nearest neighbor rescaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>area</samp>&rsquo;</span></dt>
<dd><p>Select averaging area rescaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>bicublin</samp>&rsquo;</span></dt>
<dd><p>Select bicubic scaling algorithm for the luma component, bilinear for
chroma components.
</p>
</dd>
<dt><span>&lsquo;<samp>gauss</samp>&rsquo;</span></dt>
<dd><p>Select Gaussian rescaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>sinc</samp>&rsquo;</span></dt>
<dd><p>Select sinc rescaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>lanczos</samp>&rsquo;</span></dt>
<dd><p>Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be
changed by setting <code>param0</code>.
</p>
</dd>
<dt><span>&lsquo;<samp>spline</samp>&rsquo;</span></dt>
<dd><p>Select natural bicubic spline rescaling algorithm.
</p>
</dd>
<dt><span>&lsquo;<samp>print_info</samp>&rsquo;</span></dt>
<dd><p>Enable printing/debug logging.
</p>
</dd>
<dt><span>&lsquo;<samp>accurate_rnd</samp>&rsquo;</span></dt>
<dd><p>Enable accurate rounding.
</p>
</dd>
<dt><span>&lsquo;<samp>full_chroma_int</samp>&rsquo;</span></dt>
<dd><p>Enable full chroma interpolation.
</p>
</dd>
<dt><span>&lsquo;<samp>full_chroma_inp</samp>&rsquo;</span></dt>
<dd><p>Select full chroma input.
</p>
</dd>
<dt><span>&lsquo;<samp>bitexact</samp>&rsquo;</span></dt>
<dd><p>Enable bitexact output.
</p></dd>
</dl>

</dd>
<dt><span><samp>srcw <var>(API only)</var></samp></span></dt>
<dd><p>Set source width.
</p>
</dd>
<dt><span><samp>srch <var>(API only)</var></samp></span></dt>
<dd><p>Set source height.
</p>
</dd>
<dt><span><samp>dstw <var>(API only)</var></samp></span></dt>
<dd><p>Set destination width.
</p>
</dd>
<dt><span><samp>dsth <var>(API only)</var></samp></span></dt>
<dd><p>Set destination height.
</p>
</dd>
<dt><span><samp>src_format <var>(API only)</var></samp></span></dt>
<dd><p>Set source pixel format (must be expressed as an integer).
</p>
</dd>
<dt><span><samp>dst_format <var>(API only)</var></samp></span></dt>
<dd><p>Set destination pixel format (must be expressed as an integer).
</p>
</dd>
<dt><span><samp>src_range <var>(boolean)</var></samp></span></dt>
<dd><p>If value is set to <code>1</code>, indicates source is full range. Default value is
<code>0</code>, which indicates source is limited range.
</p>
</dd>
<dt><span><samp>dst_range <var>(boolean)</var></samp></span></dt>
<dd><p>If value is set to <code>1</code>, enable full range for destination. Default value
is <code>0</code>, which enables limited range.
</p>
<span id="sws_005fparams"></span></dd>
<dt><span><samp>param0, param1</samp></span></dt>
<dd><p>Set scaling algorithm parameters. The specified values are specific of
some scaling algorithms and ignored by others. The specified values
are floating point number values.
</p>
</dd>
<dt><span><samp>sws_dither</samp></span></dt>
<dd><p>Set the dithering algorithm. Accepts one of the following
values. Default value is &lsquo;<samp>auto</samp>&rsquo;.
</p>
<dl compact="compact">
<dt><span>&lsquo;<samp>auto</samp>&rsquo;</span></dt>
<dd><p>automatic choice
</p>
</dd>
<dt><span>&lsquo;<samp>none</samp>&rsquo;</span></dt>
<dd><p>no dithering
</p>
</dd>
<dt><span>&lsquo;<samp>bayer</samp>&rsquo;</span></dt>
<dd><p>bayer dither
</p>
</dd>
<dt><span>&lsquo;<samp>ed</samp>&rsquo;</span></dt>
<dd><p>error diffusion dither
</p>
</dd>
<dt><span>&lsquo;<samp>a_dither</samp>&rsquo;</span></dt>
<dd><p>arithmetic dither, based using addition
</p>
</dd>
<dt><span>&lsquo;<samp>x_dither</samp>&rsquo;</span></dt>
<dd><p>arithmetic dither, based using xor (more random/less apparent patterning that
a_dither).
</p>
</dd>
</dl>

</dd>
<dt><span><samp>alphablend</samp></span></dt>
<dd><p>Set the alpha blending to use when the input has alpha but the output does not.
Default value is &lsquo;<samp>none</samp>&rsquo;.
</p>
<dl compact="compact">
<dt><span>&lsquo;<samp>uniform_color</samp>&rsquo;</span></dt>
<dd><p>Blend onto a uniform background color
</p>
</dd>
<dt><span>&lsquo;<samp>checkerboard</samp>&rsquo;</span></dt>
<dd><p>Blend onto a checkerboard
</p>
</dd>
<dt><span>&lsquo;<samp>none</samp>&rsquo;</span></dt>
<dd><p>No blending
</p>
</dd>
</dl>

</dd>
</dl>


<a name="See-Also"></a>
<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="libswscale.html">libswscale</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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ffplay Documentation
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ffplay Documentation
</h1>
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Synopsis" href="#Synopsis">1 Synopsis</a></li>
<li><a id="toc-Description" href="#Description">2 Description</a></li>
<li><a id="toc-Options" href="#Options">3 Options</a>
<ul class="no-bullet">
<li><a id="toc-Stream-specifiers-1" href="#Stream-specifiers-1">3.1 Stream specifiers</a></li>
<li><a id="toc-Generic-options" href="#Generic-options">3.2 Generic options</a></li>
<li><a id="toc-AVOptions" href="#AVOptions">3.3 AVOptions</a></li>
<li><a id="toc-Main-options" href="#Main-options">3.4 Main options</a></li>
<li><a id="toc-Advanced-options" href="#Advanced-options">3.5 Advanced options</a></li>
<li><a id="toc-While-playing" href="#While-playing">3.6 While playing</a></li>
</ul></li>
<li><a id="toc-See-Also" href="#See-Also">4 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">5 Authors</a></li>
</ul>
</div>
</div>

<a name="Synopsis"></a>
<h2 class="chapter">1 Synopsis<span class="pull-right"><a class="anchor hidden-xs" href="#Synopsis" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Synopsis" aria-hidden="true">TOC</a></span></h2>

<p>ffplay [<var>options</var>] [<samp>input_url</samp>]
</p>
<a name="Description"></a>
<h2 class="chapter">2 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>FFplay is a very simple and portable media player using the FFmpeg
libraries and the SDL library. It is mostly used as a testbed for the
various FFmpeg APIs.
</p>
<a name="Options"></a>
<h2 class="chapter">3 Options<span class="pull-right"><a class="anchor hidden-xs" href="#Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Options" aria-hidden="true">TOC</a></span></h2>

<p>All the numerical options, if not specified otherwise, accept a string
representing a number as input, which may be followed by one of the SI
unit prefixes, for example: &rsquo;K&rsquo;, &rsquo;M&rsquo;, or &rsquo;G&rsquo;.
</p>
<p>If &rsquo;i&rsquo; is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
powers of 1024 instead of powers of 1000. Appending &rsquo;B&rsquo; to the SI unit
prefix multiplies the value by 8. This allows using, for example:
&rsquo;KB&rsquo;, &rsquo;MiB&rsquo;, &rsquo;G&rsquo; and &rsquo;B&rsquo; as number suffixes.
</p>
<p>Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
the option name with &quot;no&quot;. For example using &quot;-nofoo&quot;
will set the boolean option with name &quot;foo&quot; to false.
</p>
<span id="Stream-specifiers"></span><a name="Stream-specifiers-1"></a>
<h3 class="section">3.1 Stream specifiers<span class="pull-right"><a class="anchor hidden-xs" href="#Stream-specifiers-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Stream-specifiers-1" aria-hidden="true">TOC</a></span></h3>
<p>Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) a given option belongs to.
</p>
<p>A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. <code>-codec:a:1 ac3</code> contains the
<code>a:1</code> stream specifier, which matches the second audio stream. Therefore, it
would select the ac3 codec for the second audio stream.
</p>
<p>A stream specifier can match several streams, so that the option is applied to all
of them. E.g. the stream specifier in <code>-b:a 128k</code> matches all audio
streams.
</p>
<p>An empty stream specifier matches all streams. For example, <code>-codec copy</code>
or <code>-codec: copy</code> would copy all the streams without reencoding.
</p>
<p>Possible forms of stream specifiers are:
</p><dl compact="compact">
<dt><span><samp><var>stream_index</var></samp></span></dt>
<dd><p>Matches the stream with this index. E.g. <code>-threads:1 4</code> would set the
thread count for the second stream to 4. If <var>stream_index</var> is used as an
additional stream specifier (see below), then it selects stream number
<var>stream_index</var> from the matching streams. Stream numbering is based on the
order of the streams as detected by libavformat except when a program ID is
also specified. In this case it is based on the ordering of the streams in the
program.
</p></dd>
<dt><span><samp><var>stream_type</var>[:<var>additional_stream_specifier</var>]</samp></span></dt>
<dd><p><var>stream_type</var> is one of following: &rsquo;v&rsquo; or &rsquo;V&rsquo; for video, &rsquo;a&rsquo; for audio, &rsquo;s&rsquo;
for subtitle, &rsquo;d&rsquo; for data, and &rsquo;t&rsquo; for attachments. &rsquo;v&rsquo; matches all video
streams, &rsquo;V&rsquo; only matches video streams which are not attached pictures, video
thumbnails or cover arts. If <var>additional_stream_specifier</var> is used, then
it matches streams which both have this type and match the
<var>additional_stream_specifier</var>. Otherwise, it matches all streams of the
specified type.
</p></dd>
<dt><span><samp>p:<var>program_id</var>[:<var>additional_stream_specifier</var>]</samp></span></dt>
<dd><p>Matches streams which are in the program with the id <var>program_id</var>. If
<var>additional_stream_specifier</var> is used, then it matches streams which both
are part of the program and match the <var>additional_stream_specifier</var>.
</p>
</dd>
<dt><span><samp>#<var>stream_id</var> or i:<var>stream_id</var></samp></span></dt>
<dd><p>Match the stream by stream id (e.g. PID in MPEG-TS container).
</p></dd>
<dt><span><samp>m:<var>key</var>[:<var>value</var>]</samp></span></dt>
<dd><p>Matches streams with the metadata tag <var>key</var> having the specified value. If
<var>value</var> is not given, matches streams that contain the given tag with any
value.
</p></dd>
<dt><span><samp>u</samp></span></dt>
<dd><p>Matches streams with usable configuration, the codec must be defined and the
essential information such as video dimension or audio sample rate must be present.
</p>
<p>Note that in <code>ffmpeg</code>, matching by metadata will only work properly for
input files.
</p></dd>
</dl>

<a name="Generic-options"></a>
<h3 class="section">3.2 Generic options<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-options" aria-hidden="true">TOC</a></span></h3>

<p>These options are shared amongst the ff* tools.
</p>
<dl compact="compact">
<dt><span><samp>-L</samp></span></dt>
<dd><p>Show license.
</p>
</dd>
<dt><span><samp>-h, -?, -help, --help [<var>arg</var>]</samp></span></dt>
<dd><p>Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
</p>
<p>Possible values of <var>arg</var> are:
</p><dl compact="compact">
<dt><span><samp>long</samp></span></dt>
<dd><p>Print advanced tool options in addition to the basic tool options.
</p>
</dd>
<dt><span><samp>full</samp></span></dt>
<dd><p>Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
</p>
</dd>
<dt><span><samp>decoder=<var>decoder_name</var></samp></span></dt>
<dd><p>Print detailed information about the decoder named <var>decoder_name</var>. Use the
<samp>-decoders</samp> option to get a list of all decoders.
</p>
</dd>
<dt><span><samp>encoder=<var>encoder_name</var></samp></span></dt>
<dd><p>Print detailed information about the encoder named <var>encoder_name</var>. Use the
<samp>-encoders</samp> option to get a list of all encoders.
</p>
</dd>
<dt><span><samp>demuxer=<var>demuxer_name</var></samp></span></dt>
<dd><p>Print detailed information about the demuxer named <var>demuxer_name</var>. Use the
<samp>-formats</samp> option to get a list of all demuxers and muxers.
</p>
</dd>
<dt><span><samp>muxer=<var>muxer_name</var></samp></span></dt>
<dd><p>Print detailed information about the muxer named <var>muxer_name</var>. Use the
<samp>-formats</samp> option to get a list of all muxers and demuxers.
</p>
</dd>
<dt><span><samp>filter=<var>filter_name</var></samp></span></dt>
<dd><p>Print detailed information about the filter named <var>filter_name</var>. Use the
<samp>-filters</samp> option to get a list of all filters.
</p>
</dd>
<dt><span><samp>bsf=<var>bitstream_filter_name</var></samp></span></dt>
<dd><p>Print detailed information about the bitstream filter named <var>bitstream_filter_name</var>.
Use the <samp>-bsfs</samp> option to get a list of all bitstream filters.
</p>
</dd>
<dt><span><samp>protocol=<var>protocol_name</var></samp></span></dt>
<dd><p>Print detailed information about the protocol named <var>protocol_name</var>.
Use the <samp>-protocols</samp> option to get a list of all protocols.
</p></dd>
</dl>

</dd>
<dt><span><samp>-version</samp></span></dt>
<dd><p>Show version.
</p>
</dd>
<dt><span><samp>-buildconf</samp></span></dt>
<dd><p>Show the build configuration, one option per line.
</p>
</dd>
<dt><span><samp>-formats</samp></span></dt>
<dd><p>Show available formats (including devices).
</p>
</dd>
<dt><span><samp>-demuxers</samp></span></dt>
<dd><p>Show available demuxers.
</p>
</dd>
<dt><span><samp>-muxers</samp></span></dt>
<dd><p>Show available muxers.
</p>
</dd>
<dt><span><samp>-devices</samp></span></dt>
<dd><p>Show available devices.
</p>
</dd>
<dt><span><samp>-codecs</samp></span></dt>
<dd><p>Show all codecs known to libavcodec.
</p>
<p>Note that the term &rsquo;codec&rsquo; is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
</p>
</dd>
<dt><span><samp>-decoders</samp></span></dt>
<dd><p>Show available decoders.
</p>
</dd>
<dt><span><samp>-encoders</samp></span></dt>
<dd><p>Show all available encoders.
</p>
</dd>
<dt><span><samp>-bsfs</samp></span></dt>
<dd><p>Show available bitstream filters.
</p>
</dd>
<dt><span><samp>-protocols</samp></span></dt>
<dd><p>Show available protocols.
</p>
</dd>
<dt><span><samp>-filters</samp></span></dt>
<dd><p>Show available libavfilter filters.
</p>
</dd>
<dt><span><samp>-pix_fmts</samp></span></dt>
<dd><p>Show available pixel formats.
</p>
</dd>
<dt><span><samp>-sample_fmts</samp></span></dt>
<dd><p>Show available sample formats.
</p>
</dd>
<dt><span><samp>-layouts</samp></span></dt>
<dd><p>Show channel names and standard channel layouts.
</p>
</dd>
<dt><span><samp>-dispositions</samp></span></dt>
<dd><p>Show stream dispositions.
</p>
</dd>
<dt><span><samp>-colors</samp></span></dt>
<dd><p>Show recognized color names.
</p>
</dd>
<dt><span><samp>-sources <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></span></dt>
<dd><p>Show autodetected sources of the input device.
Some devices may provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
</p><div class="example">
<pre class="example">ffmpeg -sources pulse,server=192.168.0.4
</pre></div>

</dd>
<dt><span><samp>-sinks <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></span></dt>
<dd><p>Show autodetected sinks of the output device.
Some devices may provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
</p><div class="example">
<pre class="example">ffmpeg -sinks pulse,server=192.168.0.4
</pre></div>

</dd>
<dt><span><samp>-loglevel [<var>flags</var>+]<var>loglevel</var> | -v [<var>flags</var>+]<var>loglevel</var></samp></span></dt>
<dd><p>Set logging level and flags used by the library.
</p>
<p>The optional <var>flags</var> prefix can consist of the following values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>repeat</samp>&rsquo;</span></dt>
<dd><p>Indicates that repeated log output should not be compressed to the first line
and the &quot;Last message repeated n times&quot; line will be omitted.
</p></dd>
<dt><span>&lsquo;<samp>level</samp>&rsquo;</span></dt>
<dd><p>Indicates that log output should add a <code>[level]</code> prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
</p></dd>
</dl>
<p>Flags can also be used alone by adding a &rsquo;+&rsquo;/&rsquo;-&rsquo; prefix to set/reset a single
flag without affecting other <var>flags</var> or changing <var>loglevel</var>. When
setting both <var>flags</var> and <var>loglevel</var>, a &rsquo;+&rsquo; separator is expected
between the last <var>flags</var> value and before <var>loglevel</var>.
</p>
<p><var>loglevel</var> is a string or a number containing one of the following values:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>quiet, -8</samp>&rsquo;</span></dt>
<dd><p>Show nothing at all; be silent.
</p></dd>
<dt><span>&lsquo;<samp>panic, 0</samp>&rsquo;</span></dt>
<dd><p>Only show fatal errors which could lead the process to crash, such as
an assertion failure. This is not currently used for anything.
</p></dd>
<dt><span>&lsquo;<samp>fatal, 8</samp>&rsquo;</span></dt>
<dd><p>Only show fatal errors. These are errors after which the process absolutely
cannot continue.
</p></dd>
<dt><span>&lsquo;<samp>error, 16</samp>&rsquo;</span></dt>
<dd><p>Show all errors, including ones which can be recovered from.
</p></dd>
<dt><span>&lsquo;<samp>warning, 24</samp>&rsquo;</span></dt>
<dd><p>Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
</p></dd>
<dt><span>&lsquo;<samp>info, 32</samp>&rsquo;</span></dt>
<dd><p>Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
</p></dd>
<dt><span>&lsquo;<samp>verbose, 40</samp>&rsquo;</span></dt>
<dd><p>Same as <code>info</code>, except more verbose.
</p></dd>
<dt><span>&lsquo;<samp>debug, 48</samp>&rsquo;</span></dt>
<dd><p>Show everything, including debugging information.
</p></dd>
<dt><span>&lsquo;<samp>trace, 56</samp>&rsquo;</span></dt>
</dl>

<p>For example to enable repeated log output, add the <code>level</code> prefix, and set
<var>loglevel</var> to <code>verbose</code>:
</p><div class="example">
<pre class="example">ffmpeg -loglevel repeat+level+verbose -i input output
</pre></div>
<p>Another example that enables repeated log output without affecting current
state of <code>level</code> prefix flag or <var>loglevel</var>:
</p><div class="example">
<pre class="example">ffmpeg [...] -loglevel +repeat
</pre></div>

<p>By default the program logs to stderr. If coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
<code>AV_LOG_FORCE_NOCOLOR</code>, or can be forced setting
the environment variable <code>AV_LOG_FORCE_COLOR</code>.
</p>
</dd>
<dt><span><samp>-report</samp></span></dt>
<dd><p>Dump full command line and log output to a file named
<code><var>program</var>-<var>YYYYMMDD</var>-<var>HHMMSS</var>.log</code> in the current
directory.
This file can be useful for bug reports.
It also implies <code>-loglevel debug</code>.
</p>
<p>Setting the environment variable <code>FFREPORT</code> to any value has the
same effect. If the value is a &rsquo;:&rsquo;-separated key=value sequence, these
options will affect the report; option values must be escaped if they
contain special characters or the options delimiter &rsquo;:&rsquo; (see the
&ldquo;Quoting and escaping&rdquo; section in the ffmpeg-utils manual).
</p>
<p>The following options are recognized:
</p><dl compact="compact">
<dt><span><samp>file</samp></span></dt>
<dd><p>set the file name to use for the report; <code>%p</code> is expanded to the name
of the program, <code>%t</code> is expanded to a timestamp, <code>%%</code> is expanded
to a plain <code>%</code>
</p></dd>
<dt><span><samp>level</samp></span></dt>
<dd><p>set the log verbosity level using a numerical value (see <code>-loglevel</code>).
</p></dd>
</dl>

<p>For example, to output a report to a file named <samp>ffreport.log</samp>
using a log level of <code>32</code> (alias for log level <code>info</code>):
</p>
<div class="example">
<pre class="example">FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
</pre></div>

<p>Errors in parsing the environment variable are not fatal, and will not
appear in the report.
</p>
</dd>
<dt><span><samp>-hide_banner</samp></span></dt>
<dd><p>Suppress printing banner.
</p>
<p>All FFmpeg tools will normally show a copyright notice, build options
and library versions. This option can be used to suppress printing
this information.
</p>
</dd>
<dt><span><samp>-cpuflags flags (<em>global</em>)</samp></span></dt>
<dd><p>Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you&rsquo;re doing.
</p><div class="example">
<pre class="example">ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
</pre></div>
<p>Possible flags for this option are:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>x86</samp>&rsquo;</span></dt>
<dd><dl compact="compact">
<dt><span>&lsquo;<samp>mmx</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>mmxext</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse2slow</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse3</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse3slow</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>ssse3</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>atom</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse4.1</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>sse4.2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>avx</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>avx2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>xop</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>fma3</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>fma4</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>3dnow</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>3dnowext</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>bmi1</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>bmi2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>cmov</samp>&rsquo;</span></dt>
</dl>
</dd>
<dt><span>&lsquo;<samp>ARM</samp>&rsquo;</span></dt>
<dd><dl compact="compact">
<dt><span>&lsquo;<samp>armv5te</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>armv6</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>armv6t2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>vfp</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>vfpv3</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>neon</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>setend</samp>&rsquo;</span></dt>
</dl>
</dd>
<dt><span>&lsquo;<samp>AArch64</samp>&rsquo;</span></dt>
<dd><dl compact="compact">
<dt><span>&lsquo;<samp>armv8</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>vfp</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>neon</samp>&rsquo;</span></dt>
</dl>
</dd>
<dt><span>&lsquo;<samp>PowerPC</samp>&rsquo;</span></dt>
<dd><dl compact="compact">
<dt><span>&lsquo;<samp>altivec</samp>&rsquo;</span></dt>
</dl>
</dd>
<dt><span>&lsquo;<samp>Specific Processors</samp>&rsquo;</span></dt>
<dd><dl compact="compact">
<dt><span>&lsquo;<samp>pentium2</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>pentium3</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>pentium4</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>k6</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>k62</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>athlon</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>athlonxp</samp>&rsquo;</span></dt>
<dt><span>&lsquo;<samp>k8</samp>&rsquo;</span></dt>
</dl>
</dd>
</dl>

</dd>
<dt><span><samp>-cpucount <var>count</var> (<em>global</em>)</samp></span></dt>
<dd><p>Override detection of CPU count. This option is intended
for testing. Do not use it unless you know what you&rsquo;re doing.
</p><div class="example">
<pre class="example">ffmpeg -cpucount 2
</pre></div>

</dd>
<dt><span><samp>-max_alloc <var>bytes</var></samp></span></dt>
<dd><p>Set the maximum size limit for allocating a block on the heap by ffmpeg&rsquo;s
family of malloc functions. Exercise <strong>extreme caution</strong> when using
this option. Don&rsquo;t use if you do not understand the full consequence of doing so.
Default is INT_MAX.
</p></dd>
</dl>

<a name="AVOptions"></a>
<h3 class="section">3.3 AVOptions<span class="pull-right"><a class="anchor hidden-xs" href="#AVOptions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVOptions" aria-hidden="true">TOC</a></span></h3>

<p>These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
<samp>-help</samp> option. They are separated into two categories:
</p><dl compact="compact">
<dt><span><samp>generic</samp></span></dt>
<dd><p>These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
</p></dd>
<dt><span><samp>private</samp></span></dt>
<dd><p>These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
</p></dd>
</dl>

<p>For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the <samp>id3v2_version</samp> private option of the MP3
muxer:
</p><div class="example">
<pre class="example">ffmpeg -i input.flac -id3v2_version 3 out.mp3
</pre></div>

<p>All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them:
</p><div class="example">
<pre class="example">ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
</pre></div>

<p>In the above example, a multichannel audio stream is mapped twice for output.
The first instance is encoded with codec ac3 and bitrate 640k.
The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using
absolute index of the output stream.
</p>
<p>Note: the <samp>-nooption</samp> syntax cannot be used for boolean
AVOptions, use <samp>-option 0</samp>/<samp>-option 1</samp>.
</p>
<p>Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.
</p>
<a name="Main-options"></a>
<h3 class="section">3.4 Main options<span class="pull-right"><a class="anchor hidden-xs" href="#Main-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Main-options" aria-hidden="true">TOC</a></span></h3>

<dl compact="compact">
<dt><span><samp>-x <var>width</var></samp></span></dt>
<dd><p>Force displayed width.
</p></dd>
<dt><span><samp>-y <var>height</var></samp></span></dt>
<dd><p>Force displayed height.
</p></dd>
<dt><span><samp>-fs</samp></span></dt>
<dd><p>Start in fullscreen mode.
</p></dd>
<dt><span><samp>-an</samp></span></dt>
<dd><p>Disable audio.
</p></dd>
<dt><span><samp>-vn</samp></span></dt>
<dd><p>Disable video.
</p></dd>
<dt><span><samp>-sn</samp></span></dt>
<dd><p>Disable subtitles.
</p></dd>
<dt><span><samp>-ss <var>pos</var></samp></span></dt>
<dd><p>Seek to <var>pos</var>. Note that in most formats it is not possible to seek
exactly, so <code>ffplay</code> will seek to the nearest seek point to
<var>pos</var>.
</p>
<p><var>pos</var> must be a time duration specification,
see <a data-manual="ffmpeg-utils" href="ffmpeg-utils.html#time-duration-syntax">(ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual</a>.
</p></dd>
<dt><span><samp>-t <var>duration</var></samp></span></dt>
<dd><p>Play <var>duration</var> seconds of audio/video.
</p>
<p><var>duration</var> must be a time duration specification,
see <a data-manual="ffmpeg-utils" href="ffmpeg-utils.html#time-duration-syntax">(ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual</a>.
</p></dd>
<dt><span><samp>-bytes</samp></span></dt>
<dd><p>Seek by bytes.
</p></dd>
<dt><span><samp>-seek_interval</samp></span></dt>
<dd><p>Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
</p></dd>
<dt><span><samp>-nodisp</samp></span></dt>
<dd><p>Disable graphical display.
</p></dd>
<dt><span><samp>-noborder</samp></span></dt>
<dd><p>Borderless window.
</p></dd>
<dt><span><samp>-alwaysontop</samp></span></dt>
<dd><p>Window always on top. Available on: X11 with SDL &gt;= 2.0.5, Windows SDL &gt;= 2.0.6.
</p></dd>
<dt><span><samp>-volume</samp></span></dt>
<dd><p>Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
as 100.
</p></dd>
<dt><span><samp>-f <var>fmt</var></samp></span></dt>
<dd><p>Force format.
</p></dd>
<dt><span><samp>-window_title <var>title</var></samp></span></dt>
<dd><p>Set window title (default is the input filename).
</p></dd>
<dt><span><samp>-left <var>title</var></samp></span></dt>
<dd><p>Set the x position for the left of the window (default is a centered window).
</p></dd>
<dt><span><samp>-top <var>title</var></samp></span></dt>
<dd><p>Set the y position for the top of the window (default is a centered window).
</p></dd>
<dt><span><samp>-loop <var>number</var></samp></span></dt>
<dd><p>Loops movie playback &lt;number&gt; times. 0 means forever.
</p></dd>
<dt><span><samp>-showmode <var>mode</var></samp></span></dt>
<dd><p>Set the show mode to use.
Available values for <var>mode</var> are:
</p><dl compact="compact">
<dt><span>&lsquo;<samp>0, video</samp>&rsquo;</span></dt>
<dd><p>show video
</p></dd>
<dt><span>&lsquo;<samp>1, waves</samp>&rsquo;</span></dt>
<dd><p>show audio waves
</p></dd>
<dt><span>&lsquo;<samp>2, rdft</samp>&rsquo;</span></dt>
<dd><p>show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
</p></dd>
</dl>

<p>Default value is &quot;video&quot;, if video is not present or cannot be played
&quot;rdft&quot; is automatically selected.
</p>
<p>You can interactively cycle through the available show modes by
pressing the key <tt class="key">w</tt>.
</p>
</dd>
<dt><span><samp>-vf <var>filtergraph</var></samp></span></dt>
<dd><p>Create the filtergraph specified by <var>filtergraph</var> and use it to
filter the video stream.
</p>
<p><var>filtergraph</var> is a description of the filtergraph to apply to
the stream, and must have a single video input and a single video
output. In the filtergraph, the input is associated to the label
<code>in</code>, and the output to the label <code>out</code>. See the
ffmpeg-filters manual for more information about the filtergraph
syntax.
</p>
<p>You can specify this parameter multiple times and cycle through the specified
filtergraphs along with the show modes by pressing the key <tt class="key">w</tt>.
</p>
</dd>
<dt><span><samp>-af <var>filtergraph</var></samp></span></dt>
<dd><p><var>filtergraph</var> is a description of the filtergraph to apply to
the input audio.
Use the option &quot;-filters&quot; to show all the available filters (including
sources and sinks).
</p>
</dd>
<dt><span><samp>-i <var>input_url</var></samp></span></dt>
<dd><p>Read <var>input_url</var>.
</p></dd>
</dl>

<a name="Advanced-options"></a>
<h3 class="section">3.5 Advanced options<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-options" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><span><samp>-stats</samp></span></dt>
<dd><p>Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is shown by default, unless the
log level is lower than <code>info</code>. Its display can be forced by manually
specifying this option. To disable it, you need to specify <code>-nostats</code>.
</p>
</dd>
<dt><span><samp>-fast</samp></span></dt>
<dd><p>Non-spec-compliant optimizations.
</p></dd>
<dt><span><samp>-genpts</samp></span></dt>
<dd><p>Generate pts.
</p></dd>
<dt><span><samp>-sync <var>type</var></samp></span></dt>
<dd><p>Set the master clock to audio (<code>type=audio</code>), video
(<code>type=video</code>) or external (<code>type=ext</code>). Default is audio. The
master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
</p></dd>
<dt><span><samp>-ast <var>audio_stream_specifier</var></samp></span></dt>
<dd><p>Select the desired audio stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; audio stream is selected in the program of the
already selected video stream.
</p></dd>
<dt><span><samp>-vst <var>video_stream_specifier</var></samp></span></dt>
<dd><p>Select the desired video stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; video stream is selected.
</p></dd>
<dt><span><samp>-sst <var>subtitle_stream_specifier</var></samp></span></dt>
<dd><p>Select the desired subtitle stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; subtitle stream is selected in the program of the
already selected video or audio stream.
</p></dd>
<dt><span><samp>-autoexit</samp></span></dt>
<dd><p>Exit when video is done playing.
</p></dd>
<dt><span><samp>-exitonkeydown</samp></span></dt>
<dd><p>Exit if any key is pressed.
</p></dd>
<dt><span><samp>-exitonmousedown</samp></span></dt>
<dd><p>Exit if any mouse button is pressed.
</p>
</dd>
<dt><span><samp>-codec:<var>media_specifier</var> <var>codec_name</var></samp></span></dt>
<dd><p>Force a specific decoder implementation for the stream identified by
<var>media_specifier</var>, which can assume the values <code>a</code> (audio),
<code>v</code> (video), and <code>s</code> subtitle.
</p>
</dd>
<dt><span><samp>-acodec <var>codec_name</var></samp></span></dt>
<dd><p>Force a specific audio decoder.
</p>
</dd>
<dt><span><samp>-vcodec <var>codec_name</var></samp></span></dt>
<dd><p>Force a specific video decoder.
</p>
</dd>
<dt><span><samp>-scodec <var>codec_name</var></samp></span></dt>
<dd><p>Force a specific subtitle decoder.
</p>
</dd>
<dt><span><samp>-autorotate</samp></span></dt>
<dd><p>Automatically rotate the video according to file metadata. Enabled by
default, use <samp>-noautorotate</samp> to disable it.
</p>
</dd>
<dt><span><samp>-framedrop</samp></span></dt>
<dd><p>Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use <samp>-noframedrop</samp> to disable it.
</p>
</dd>
<dt><span><samp>-infbuf</samp></span></dt>
<dd><p>Do not limit the input buffer size, read as much data as possible from the
input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use <samp>-noinfbuf</samp> to disable it.
</p>
</dd>
<dt><span><samp>-filter_threads <var>nb_threads</var></samp></span></dt>
<dd><p>Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
</p>
</dd>
</dl>

<a name="While-playing"></a>
<h3 class="section">3.6 While playing<span class="pull-right"><a class="anchor hidden-xs" href="#While-playing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-While-playing" aria-hidden="true">TOC</a></span></h3>

<dl compact="compact">
<dt><span><tt class="key">q, ESC</tt></span></dt>
<dd><p>Quit.
</p>
</dd>
<dt><span><tt class="key">f</tt></span></dt>
<dd><p>Toggle full screen.
</p>
</dd>
<dt><span><tt class="key">p, SPC</tt></span></dt>
<dd><p>Pause.
</p>
</dd>
<dt><span><tt class="key">m</tt></span></dt>
<dd><p>Toggle mute.
</p>
</dd>
<dt><span><tt class="key">9, 0</tt></span></dt>
<dt><span><tt class="key">/, *</tt></span></dt>
<dd><p>Decrease and increase volume respectively.
</p>
</dd>
<dt><span><tt class="key">a</tt></span></dt>
<dd><p>Cycle audio channel in the current program.
</p>
</dd>
<dt><span><tt class="key">v</tt></span></dt>
<dd><p>Cycle video channel.
</p>
</dd>
<dt><span><tt class="key">t</tt></span></dt>
<dd><p>Cycle subtitle channel in the current program.
</p>
</dd>
<dt><span><tt class="key">c</tt></span></dt>
<dd><p>Cycle program.
</p>
</dd>
<dt><span><tt class="key">w</tt></span></dt>
<dd><p>Cycle video filters or show modes.
</p>
</dd>
<dt><span><tt class="key">s</tt></span></dt>
<dd><p>Step to the next frame.
</p>
<p>Pause if the stream is not already paused, step to the next video
frame, and pause.
</p>
</dd>
<dt><span><tt class="key">left/right</tt></span></dt>
<dd><p>Seek backward/forward 10 seconds.
</p>
</dd>
<dt><span><tt class="key">down/up</tt></span></dt>
<dd><p>Seek backward/forward 1 minute.
</p>
</dd>
<dt><span><tt class="key">page down/page up</tt></span></dt>
<dd><p>Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
</p>
</dd>
<dt><span><tt class="key">right mouse click</tt></span></dt>
<dd><p>Seek to percentage in file corresponding to fraction of width.
</p>
</dd>
<dt><span><tt class="key">left mouse double-click</tt></span></dt>
<dd><p>Toggle full screen.
</p>
</dd>
</dl>



<a name="See-Also"></a>
<h2 class="chapter">4 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffplay-all.html">ffmpeg-all</a>,
<a href="ffmpeg.html">ffmpeg</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-utils.html">ffmpeg-utils</a>,
<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>,
<a href="ffmpeg-bitstream-filters.html">ffmpeg-bitstream-filters</a>,
<a href="ffmpeg-formats.html">ffmpeg-formats</a>,
<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
<a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
<a href="ffmpeg-filters.html">ffmpeg-filters</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">5 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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Using Git to develop FFmpeg
</title>
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<h1>
Using Git to develop FFmpeg
</h1>
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Introduction" href="#Introduction">1 Introduction</a></li>
<li><a id="toc-Basics-Usage" href="#Basics-Usage">2 Basics Usage</a>
<ul class="no-bullet">
<li><a id="toc-Get-Git" href="#Get-Git">2.1 Get Git</a></li>
<li><a id="toc-Cloning-the-source-tree" href="#Cloning-the-source-tree">2.2 Cloning the source tree</a></li>
<li><a id="toc-Updating-the-source-tree-to-the-latest-revision-1" href="#Updating-the-source-tree-to-the-latest-revision-1">2.3 Updating the source tree to the latest revision</a></li>
<li><a id="toc-Rebasing-your-local-branches" href="#Rebasing-your-local-branches">2.4 Rebasing your local branches</a></li>
<li><a id="toc-Adding_002fremoving-files_002fdirectories" href="#Adding_002fremoving-files_002fdirectories">2.5 Adding/removing files/directories</a></li>
<li><a id="toc-Showing-modifications" href="#Showing-modifications">2.6 Showing modifications</a></li>
<li><a id="toc-Inspecting-the-changelog" href="#Inspecting-the-changelog">2.7 Inspecting the changelog</a></li>
<li><a id="toc-Checking-source-tree-status" href="#Checking-source-tree-status">2.8 Checking source tree status</a></li>
<li><a id="toc-Committing" href="#Committing">2.9 Committing</a></li>
<li><a id="toc-Writing-a-commit-message" href="#Writing-a-commit-message">2.10 Writing a commit message</a></li>
<li><a id="toc-Preparing-a-patchset" href="#Preparing-a-patchset">2.11 Preparing a patchset</a></li>
<li><a id="toc-Sending-patches-for-review" href="#Sending-patches-for-review">2.12 Sending patches for review</a></li>
<li><a id="toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files">2.13 Renaming/moving/copying files or contents of files</a></li>
</ul></li>
<li><a id="toc-Git-configuration" href="#Git-configuration">3 Git configuration</a>
<ul class="no-bullet">
<li><a id="toc-Personal-Git-installation" href="#Personal-Git-installation">3.1 Personal Git installation</a></li>
<li><a id="toc-Repository-configuration" href="#Repository-configuration">3.2 Repository configuration</a></li>
</ul></li>
<li><a id="toc-FFmpeg-specific" href="#FFmpeg-specific">4 FFmpeg specific</a>
<ul class="no-bullet">
<li><a id="toc-Reverting-broken-commits" href="#Reverting-broken-commits">4.1 Reverting broken commits</a></li>
<li><a id="toc-Pushing-changes-to-remote-trees" href="#Pushing-changes-to-remote-trees">4.2 Pushing changes to remote trees</a></li>
<li><a id="toc-Finding-a-specific-svn-revision" href="#Finding-a-specific-svn-revision">4.3 Finding a specific svn revision</a></li>
</ul></li>
<li><a id="toc-gpg-key-generation" href="#gpg-key-generation">5 gpg key generation</a></li>
<li><a id="toc-Pre_002dpush-checklist" href="#Pre_002dpush-checklist">6 Pre-push checklist</a></li>
<li><a id="toc-Server-Issues" href="#Server-Issues">7 Server Issues</a></li>
</ul>
</div>
</div>

<a name="Introduction"></a>
<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>

<p>This document aims in giving some quick references on a set of useful Git
commands. You should always use the extensive and detailed documentation
provided directly by Git:
</p>
<div class="example">
<pre class="example">git --help
man git
</pre></div>

<p>shows you the available subcommands,
</p>
<div class="example">
<pre class="example">git &lt;command&gt; --help
man git-&lt;command&gt;
</pre></div>

<p>shows information about the subcommand &lt;command&gt;.
</p>
<p>Additional information could be found on the
<a href="http://gitref.org">Git Reference</a> website.
</p>
<p>For more information about the Git project, visit the
<a href="http://git-scm.com/">Git website</a>.
</p>
<p>Consult these resources whenever you have problems, they are quite exhaustive.
</p>
<p>What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project.
</p>
<a name="Basics-Usage"></a>
<h2 class="chapter">2 Basics Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Basics-Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Basics-Usage" aria-hidden="true">TOC</a></span></h2>

<a name="Get-Git"></a>
<h3 class="section">2.1 Get Git<span class="pull-right"><a class="anchor hidden-xs" href="#Get-Git" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Get-Git" aria-hidden="true">TOC</a></span></h3>

<p>You can get Git from <a href="http://git-scm.com/">http://git-scm.com/</a>
Most distribution and operating system provide a package for it.
</p>

<a name="Cloning-the-source-tree"></a>
<h3 class="section">2.2 Cloning the source tree<span class="pull-right"><a class="anchor hidden-xs" href="#Cloning-the-source-tree" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cloning-the-source-tree" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git clone https://git.ffmpeg.org/ffmpeg.git &lt;target&gt;
</pre></div>

<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var>.
</p>
<div class="example">
<pre class="example">git clone git@source.ffmpeg.org:ffmpeg &lt;target&gt;
</pre></div>

<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var> and let
you push back your changes to the remote repository.
</p>
<div class="example">
<pre class="example">git clone gil@ffmpeg.org:ffmpeg-web &lt;target&gt;
</pre></div>

<p>This will put the source of the FFmpeg website into the directory
<var>&lt;target&gt;</var> and let you push back your changes to the remote repository.
(Note that <var>gil</var> stands for GItoLite and is not a typo of <var>git</var>.)
</p>
<p>If you don&rsquo;t have write-access to the ffmpeg-web repository, you can
create patches after making a read-only ffmpeg-web clone:
</p>
<div class="example">
<pre class="example">git clone git://ffmpeg.org/ffmpeg-web &lt;target&gt;
</pre></div>

<p>Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
</p>
<div class="example">
<pre class="example">git config --global core.autocrlf false
</pre></div>


<span id="Updating-the-source-tree-to-the-latest-revision"></span><a name="Updating-the-source-tree-to-the-latest-revision-1"></a>
<h3 class="section">2.3 Updating the source tree to the latest revision<span class="pull-right"><a class="anchor hidden-xs" href="#Updating-the-source-tree-to-the-latest-revision-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Updating-the-source-tree-to-the-latest-revision-1" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git pull (--rebase)
</pre></div>

<p>pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
</p>
<div class="warning">
<p><code>--rebase</code> (see below) is recommended.
</p></div>
<a name="Rebasing-your-local-branches"></a>
<h3 class="section">2.4 Rebasing your local branches<span class="pull-right"><a class="anchor hidden-xs" href="#Rebasing-your-local-branches" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Rebasing-your-local-branches" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git pull --rebase
</pre></div>

<p>fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg&rsquo;s master tree. The master tree will reject pushes with merge commits.
</p>

<a name="Adding_002fremoving-files_002fdirectories"></a>
<h3 class="section">2.5 Adding/removing files/directories<span class="pull-right"><a class="anchor hidden-xs" href="#Adding_002fremoving-files_002fdirectories" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding_002fremoving-files_002fdirectories" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git add [-A] &lt;filename/dirname&gt;
git rm [-r] &lt;filename/dirname&gt;
</pre></div>

<p>Git needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
</p>

<a name="Showing-modifications"></a>
<h3 class="section">2.6 Showing modifications<span class="pull-right"><a class="anchor hidden-xs" href="#Showing-modifications" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Showing-modifications" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git diff &lt;filename(s)&gt;
</pre></div>

<p>will show all local modifications in your working directory as unified diff.
</p>

<a name="Inspecting-the-changelog"></a>
<h3 class="section">2.7 Inspecting the changelog<span class="pull-right"><a class="anchor hidden-xs" href="#Inspecting-the-changelog" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Inspecting-the-changelog" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git log &lt;filename(s)&gt;
</pre></div>

<p>You may also use the graphical tools like <code>gitview</code> or <code>gitk</code>
or the web interface available at <a href="http://source.ffmpeg.org/">http://source.ffmpeg.org/</a>.
</p>
<a name="Checking-source-tree-status"></a>
<h3 class="section">2.8 Checking source tree status<span class="pull-right"><a class="anchor hidden-xs" href="#Checking-source-tree-status" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Checking-source-tree-status" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git status
</pre></div>

<p>detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
</p>

<a name="Committing"></a>
<h3 class="section">2.9 Committing<span class="pull-right"><a class="anchor hidden-xs" href="#Committing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Committing" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git diff --check
</pre></div>

<p>to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
</p>
<p>Every one of them has been saved from looking like a fool by this many times.
It&rsquo;s very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
</p>
<p>For cosmetics-only commits you should get (almost) empty output from
</p>
<div class="example">
<pre class="example">git diff -w -b &lt;filename(s)&gt;
</pre></div>

<p>Also check the output of
</p>
<div class="example">
<pre class="example">git status
</pre></div>

<p>to make sure you don&rsquo;t have untracked files or deletions.
</p>
<div class="example">
<pre class="example">git add [-i|-p|-A] &lt;filenames/dirnames&gt;
</pre></div>

<p>Make sure you have told Git your name, email address and GPG key
</p>
<div class="example">
<pre class="example">git config --global user.name &quot;My Name&quot;
git config --global user.email my@email.invalid
git config --global user.signingkey ABCDEF0123245
</pre></div>

<p>Enable signing all commits or use -S
</p>
<div class="example">
<pre class="example">git config --global commit.gpgsign true
</pre></div>

<p>Use <samp>--global</samp> to set the global configuration for all your Git checkouts.
</p>
<p>Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
</p>

<div class="example">
<pre class="example">git commit
</pre></div>

<p>Git will commit the selected changes to your current local branch.
</p>
<p>You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
</p>
<div class="example">
<pre class="example">git config --global core.editor
</pre></div>

<p>or set by one of the following environment variables:
<var>GIT_EDITOR</var>, <var>VISUAL</var> or <var>EDITOR</var>.
</p>
<a name="Writing-a-commit-message"></a>
<h3 class="section">2.10 Writing a commit message<span class="pull-right"><a class="anchor hidden-xs" href="#Writing-a-commit-message" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Writing-a-commit-message" aria-hidden="true">TOC</a></span></h3>

<p>Log messages should be concise but descriptive.
</p>
<p>The first line must contain the context, a colon and a very short
summary of what the commit does. Details can be added, if necessary,
separated by an empty line. These details should not exceed 60-72 characters
per line, except when containing code.
</p>
<p>Example of a good commit message:
</p>
<div class="example">
<pre class="example">avcodec/cbs: add a helper to read extradata within packet side data

Using ff_cbs_read() on the raw buffer will not parse it as extradata,
resulting in parsing errors for example when handling ISOBMFF avcC.
This helper works around that.
</pre></div>

<div class="example">
<pre class="example">ptr might be NULL
</pre></div>

<p>If the summary on the first line is not enough, in the body of the message,
explain why you made a change, what you did will be obvious from the changes
themselves most of the time. Saying just &quot;bug fix&quot; or &quot;10l&quot; is bad. Remember
that people of varying skill levels look at and educate themselves while
reading through your code. Don&rsquo;t include filenames in log messages except in
the context, Git provides that information.
</p>
<p>If the commit fixes a registered issue, state it in a separate line of the
body: <code>Fix Trac ticket #42.</code>
</p>
<p>The first line will be used to name
the patch by <code>git format-patch</code>.
</p>
<p>Common mistakes for the first line, as seen in <code>git log --oneline</code>
include: missing context at the beginning; description of what the code did
before the patch; line too long or wrapped to the second line.
</p>
<a name="Preparing-a-patchset"></a>
<h3 class="section">2.11 Preparing a patchset<span class="pull-right"><a class="anchor hidden-xs" href="#Preparing-a-patchset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Preparing-a-patchset" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git format-patch &lt;commit&gt; [-o directory]
</pre></div>

<p>will generate a set of patches for each commit between <var>&lt;commit&gt;</var> and
current <var>HEAD</var>. E.g.
</p>
<div class="example">
<pre class="example">git format-patch origin/master
</pre></div>

<p>will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
</p>
<div class="example">
<pre class="example">git format-patch -n
</pre></div>

<p>which will generate patches from last <var>n</var> commits.
By default the patches are created in the current directory.
</p>
<a name="Sending-patches-for-review"></a>
<h3 class="section">2.12 Sending patches for review<span class="pull-right"><a class="anchor hidden-xs" href="#Sending-patches-for-review" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Sending-patches-for-review" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git send-email &lt;commit list|directory&gt;
</pre></div>

<p>will send the patches created by <code>git format-patch</code> or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. <var>git-email</var>
package on Debian-based distros).
</p>

<a name="Renaming_002fmoving_002fcopying-files-or-contents-of-files"></a>
<h3 class="section">2.13 Renaming/moving/copying files or contents of files<span class="pull-right"><a class="anchor hidden-xs" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">TOC</a></span></h3>

<p>Git automatically tracks such changes, making those normal commits.
</p>
<div class="example">
<pre class="example">mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
</pre></div>


<a name="Git-configuration"></a>
<h2 class="chapter">3 Git configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Git-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Git-configuration" aria-hidden="true">TOC</a></span></h2>

<p>In order to simplify a few workflows, it is advisable to configure both
your personal Git installation and your local FFmpeg repository.
</p>
<a name="Personal-Git-installation"></a>
<h3 class="section">3.1 Personal Git installation<span class="pull-right"><a class="anchor hidden-xs" href="#Personal-Git-installation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Personal-Git-installation" aria-hidden="true">TOC</a></span></h3>

<p>Add the following to your <samp>~/.gitconfig</samp> to help <code>git send-email</code>
and <code>git format-patch</code> detect renames:
</p>
<div class="example">
<pre class="example">[diff]
renames = copy
</pre></div>

<a name="Repository-configuration"></a>
<h3 class="section">3.2 Repository configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Repository-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Repository-configuration" aria-hidden="true">TOC</a></span></h3>

<p>In order to have <code>git send-email</code> automatically send patches
to the ffmpeg-devel mailing list, add the following stanza
to <samp>/path/to/ffmpeg/repository/.git/config</samp>:
</p>
<div class="example">
<pre class="example">[sendemail]
to = ffmpeg-devel@ffmpeg.org
</pre></div>

<a name="FFmpeg-specific"></a>
<h2 class="chapter">4 FFmpeg specific<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-specific" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-specific" aria-hidden="true">TOC</a></span></h2>

<a name="Reverting-broken-commits"></a>
<h3 class="section">4.1 Reverting broken commits<span class="pull-right"><a class="anchor hidden-xs" href="#Reverting-broken-commits" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Reverting-broken-commits" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git reset &lt;commit&gt;
</pre></div>

<p><code>git reset</code> will uncommit the changes till <var>&lt;commit&gt;</var> rewriting
the current branch history.
</p>
<div class="example">
<pre class="example">git commit --amend
</pre></div>

<p>allows one to amend the last commit details quickly.
</p>
<div class="example">
<pre class="example">git rebase -i origin/master
</pre></div>

<p>will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
</p>
<div class="info">
<p><code>git reset</code>, <code>git commit --amend</code> and <code>git rebase</code>
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
</p></div>
<div class="example">
<pre class="example">git revert &lt;commit&gt;
</pre></div>

<p><code>git revert</code> will generate a revert commit. This will not make the
faulty commit disappear from the history.
</p>
<a name="Pushing-changes-to-remote-trees"></a>
<h3 class="section">4.2 Pushing changes to remote trees<span class="pull-right"><a class="anchor hidden-xs" href="#Pushing-changes-to-remote-trees" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Pushing-changes-to-remote-trees" aria-hidden="true">TOC</a></span></h3>

<div class="example">
<pre class="example">git push origin master --dry-run
</pre></div>

<p>Will simulate a push of the local master branch to the default remote
(<var>origin</var>). And list which branches and ranges or commits would have been
pushed.
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to <a href="#Updating-the-source-tree-to-the-latest-revision">Updating the source tree to the latest revision</a>.
</p>
<div class="example">
<pre class="example">git remote add &lt;name&gt; &lt;url&gt;
</pre></div>

<p>Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
</p>
<div class="example">
<pre class="example">git push &lt;remote&gt; &lt;refspec&gt;
</pre></div>

<p>Will push the changes to the <var>&lt;remote&gt;</var> repository.
Omitting <var>&lt;refspec&gt;</var> makes <code>git push</code> update all the remote
branches matching the local ones.
</p>
<a name="Finding-a-specific-svn-revision"></a>
<h3 class="section">4.3 Finding a specific svn revision<span class="pull-right"><a class="anchor hidden-xs" href="#Finding-a-specific-svn-revision" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Finding-a-specific-svn-revision" aria-hidden="true">TOC</a></span></h3>

<p>Since version 1.7.1 Git supports &lsquo;<samp>:/foo</samp>&rsquo; syntax for specifying commits
based on a regular expression. see man gitrevisions
</p>
<div class="example">
<pre class="example">git show :/'as revision 23456'
</pre></div>

<p>will show the svn changeset &lsquo;<samp>r23456</samp>&rsquo;. With older Git versions searching in
the <code>git log</code> output is the easiest option (especially if a pager with
search capabilities is used).
</p>
<p>This commit can be checked out with
</p>
<div class="example">
<pre class="example">git checkout -b svn_23456 :/'as revision 23456'
</pre></div>

<p>or for Git &lt; 1.7.1 with
</p>
<div class="example">
<pre class="example">git checkout -b svn_23456 $SHA1
</pre></div>

<p>where <var>$SHA1</var> is the commit hash from the <code>git log</code> output.
</p>

<a name="gpg-key-generation"></a>
<h2 class="chapter">5 gpg key generation<span class="pull-right"><a class="anchor hidden-xs" href="#gpg-key-generation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-gpg-key-generation" aria-hidden="true">TOC</a></span></h2>

<p>If you have no gpg key yet, we recommend that you create a ed25519 based key as it
is small, fast and secure. Especially it results in small signatures in git.
</p>
<div class="example">
<pre class="example">gpg --default-new-key-algo &quot;ed25519/cert,sign+cv25519/encr&quot; --quick-generate-key &quot;human@server.com&quot;
</pre></div>

<p>When generating a key, make sure the email specified matches the email used in git as some sites like
github consider mismatches a reason to declare such commits unverified. After generating a key you
can add it to the MAINTAINER file and upload it to a keyserver.
</p>
<a name="Pre_002dpush-checklist"></a>
<h2 class="chapter">6 Pre-push checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Pre_002dpush-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Pre_002dpush-checklist" aria-hidden="true">TOC</a></span></h2>

<p>Once you have a set of commits that you feel are ready for pushing,
work through the following checklist to doublecheck everything is in
proper order. This list tries to be exhaustive. In case you are just
pushing a typo in a comment, some of the steps may be unnecessary.
Apply your common sense, but if in doubt, err on the side of caution.
</p>
<p>First, make sure that the commits and branches you are going to push
match what you want pushed and that nothing is missing, extraneous or
wrong. You can see what will be pushed by running the git push command
with <samp>--dry-run</samp> first. And then inspecting the commits listed with
<code>git log -p 1234567..987654</code>. The <code>git status</code> command
may help in finding local changes that have been forgotten to be added.
</p>
<p>Next let the code pass through a full run of our test suite.
</p>
<ul>
<li> <code>make distclean</code>
</li><li> <code>/path/to/ffmpeg/configure</code>
</li><li> <code>make fate</code>
</li><li> if fate fails due to missing samples run <code>make fate-rsync</code> and retry
</li></ul>

<p>Make sure all your changes have been checked before pushing them, the
test suite only checks against regressions and that only to some extend. It does
obviously not check newly added features/code to be working unless you have
added a test for that (which is recommended).
</p>
<p>Also note that every single commit should pass the test suite, not just
the result of a series of patches.
</p>
<p>Once everything passed, push the changes to your public ffmpeg clone and post a
merge request to ffmpeg-devel. You can also push them directly but this is not
recommended.
</p>
<a name="Server-Issues"></a>
<h2 class="chapter">7 Server Issues<span class="pull-right"><a class="anchor hidden-xs" href="#Server-Issues" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Server-Issues" aria-hidden="true">TOC</a></span></h2>

<p>Contact the project admins at <a href="mailto:root@ffmpeg.org">root@ffmpeg.org</a> if you have technical
problems with the Git server.
</p> <p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavcodec Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libavcodec Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libavcodec library provides a generic encoding/decoding framework
and contains multiple decoders and encoders for audio, video and
subtitle streams, and several bitstream filters.
</p>
<p>The shared architecture provides various services ranging from bit
stream I/O to DSP optimizations, and makes it suitable for
implementing robust and fast codecs as well as for experimentation.
</p>

<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>, <a href="ffmpeg-bitstream-filters.html">bitstream-filters</a>,
<a href="libavutil.html">libavutil</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavdevice Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libavdevice Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libavdevice library provides a generic framework for grabbing from
and rendering to many common multimedia input/output devices, and
supports several input and output devices, including Video4Linux2,
VfW, DShow, and ALSA.
</p>

<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavfilter Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libavfilter Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libavfilter library provides a generic audio/video filtering
framework containing several filters, sources and sinks.
</p>

<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-filters.html">ffmpeg-filters</a>,
<a href="libavutil.html">libavutil</a>, <a href="libswscale.html">libswscale</a>, <a href="libswresample.html">libswresample</a>,
<a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>, <a href="libavdevice.html">libavdevice</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavformat Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libavformat Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libavformat library provides a generic framework for multiplexing
and demultiplexing (muxing and demuxing) audio, video and subtitle
streams. It encompasses multiple muxers and demuxers for multimedia
container formats.
</p>
<p>It also supports several input and output protocols to access a media
resource.
</p>

<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-formats.html">ffmpeg-formats</a>, <a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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<html>
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<head>
<meta charset="utf-8">
<title>
Libavutil Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libavutil Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats). It is not a library for
code needed by both libavcodec and libavformat.
</p>
<p>The goals for this library is to be:
</p>
<dl compact="compact">
<dt><span><strong>Modular</strong></span></dt>
<dd><p>It should have few interdependencies and the possibility of disabling individual
parts during <code>./configure</code>.
</p>
</dd>
<dt><span><strong>Small</strong></span></dt>
<dd><p>Both sources and objects should be small.
</p>
</dd>
<dt><span><strong>Efficient</strong></span></dt>
<dd><p>It should have low CPU and memory usage.
</p>
</dd>
<dt><span><strong>Useful</strong></span></dt>
<dd><p>It should avoid useless features that almost no one needs.
</p></dd>
</dl>


<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-utils.html">ffmpeg-utils</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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<html>
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<head>
<meta charset="utf-8">
<title>
Libswresample Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libswresample Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libswresample library performs highly optimized audio resampling,
rematrixing and sample format conversion operations.
</p>
<p>Specifically, this library performs the following conversions:
</p>
<ul>
<li> <em>Resampling</em>: is the process of changing the audio rate, for
example from a high sample rate of 44100Hz to 8000Hz. Audio
conversion from high to low sample rate is a lossy process. Several
resampling options and algorithms are available.

</li><li> <em>Format conversion</em>: is the process of converting the type of
samples, for example from 16-bit signed samples to unsigned 8-bit or
float samples. It also handles packing conversion, when passing from
packed layout (all samples belonging to distinct channels interleaved
in the same buffer), to planar layout (all samples belonging to the
same channel stored in a dedicated buffer or &quot;plane&quot;).

</li><li> <em>Rematrixing</em>: is the process of changing the channel layout, for
example from stereo to mono. When the input channels cannot be mapped
to the output streams, the process is lossy, since it involves
different gain factors and mixing.
</li></ul>

<p>Various other audio conversions (e.g. stretching and padding) are
enabled through dedicated options.
</p>

<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
<a href="libavutil.html">libavutil</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
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<html>
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<head>
<meta charset="utf-8">
<title>
Libswscale Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
Libswscale Documentation
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a id="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>

<p>The libswscale library performs highly optimized image scaling and
colorspace and pixel format conversion operations.
</p>
<p>Specifically, this library performs the following conversions:
</p>
<ul>
<li> <em>Rescaling</em>: is the process of changing the video size. Several
rescaling options and algorithms are available. This is usually a
lossy process.

</li><li> <em>Pixel format conversion</em>: is the process of converting the image
format and colorspace of the image, for example from planar YUV420P to
RGB24 packed. It also handles packing conversion, that is converts
from packed layout (all pixels belonging to distinct planes
interleaved in the same buffer), to planar layout (all samples
belonging to the same plane stored in a dedicated buffer or &quot;plane&quot;).

<p>This is usually a lossy process in case the source and destination
colorspaces differ.
</p></li></ul>


<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>

<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>,
<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
<a href="libavutil.html">libavutil</a>
</p>

<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>

<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<html>
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<head>
<meta charset="utf-8">
<title>
FFmpeg Mailing List FAQ
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
FFmpeg Mailing List FAQ
</h1>
<div align="center">
</div>


<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-General-Questions" href="#General-Questions">1 General Questions</a>
<ul class="no-bullet">
<li><a id="toc-What-is-a-mailing-list_003f" href="#What-is-a-mailing-list_003f">1.1 What is a mailing list?</a></li>
<li><a id="toc-What-type-of-questions-can-I-ask_003f" href="#What-type-of-questions-can-I-ask_003f">1.2 What type of questions can I ask?</a></li>
<li><a id="toc-How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f-1" href="#How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f-1">1.3 How do I ask a question or send a message to a mailing list?</a></li>
</ul></li>
<li><a id="toc-Subscribing-_002f-Unsubscribing" href="#Subscribing-_002f-Unsubscribing">2 Subscribing / Unsubscribing</a>
<ul class="no-bullet">
<li><a id="toc-How-do-I-subscribe_003f-1" href="#How-do-I-subscribe_003f-1">2.1 How do I subscribe?</a></li>
<li><a id="toc-How-do-I-unsubscribe_003f" href="#How-do-I-unsubscribe_003f">2.2 How do I unsubscribe?</a></li>
</ul></li>
<li><a id="toc-Moderation-Queue" href="#Moderation-Queue">3 Moderation Queue</a>
<ul class="no-bullet">
<li><a id="toc-Why-is-my-message-awaiting-moderator-approval_003f-1" href="#Why-is-my-message-awaiting-moderator-approval_003f-1">3.1 Why is my message awaiting moderator approval?</a></li>
<li><a id="toc-How-long-does-it-take-for-my-message-in-the-moderation-queue-to-be-approved_003f" href="#How-long-does-it-take-for-my-message-in-the-moderation-queue-to-be-approved_003f">3.2 How long does it take for my message in the moderation queue to be approved?</a></li>
<li><a id="toc-How-do-I-delete-my-message-in-the-moderation-queue_003f-1" href="#How-do-I-delete-my-message-in-the-moderation-queue_003f-1">3.3 How do I delete my message in the moderation queue?</a></li>
</ul></li>
<li><a id="toc-Archives" href="#Archives">4 Archives</a>
<ul class="no-bullet">
<li><a id="toc-Where-are-the-archives_003f-1" href="#Where-are-the-archives_003f-1">4.1 Where are the archives?</a></li>
<li><a id="toc-How-do-I-reply-to-a-message-in-the-archives_003f" href="#How-do-I-reply-to-a-message-in-the-archives_003f">4.2 How do I reply to a message in the archives?</a></li>
<li><a id="toc-How-do-I-search-the-archives_003f" href="#How-do-I-search-the-archives_003f">4.3 How do I search the archives?</a></li>
</ul></li>
<li><a id="toc-Other" href="#Other">5 Other</a>
<ul class="no-bullet">
<li><a id="toc-Is-there-an-alternative-to-the-mailing-list_003f" href="#Is-there-an-alternative-to-the-mailing-list_003f">5.1 Is there an alternative to the mailing list?</a></li>
<li><a id="toc-What-is-top_002dposting_003f-1" href="#What-is-top_002dposting_003f-1">5.2 What is top-posting?</a></li>
<li><a id="toc-What-is-the-message-size-limit_003f-1" href="#What-is-the-message-size-limit_003f-1">5.3 What is the message size limit?</a></li>
<li><a id="toc-Where-can-I-upload-sample-files_003f" href="#Where-can-I-upload-sample-files_003f">5.4 Where can I upload sample files?</a></li>
<li><a id="toc-Will-I-receive-spam-if-I-send-and_002for-subscribe-to-a-mailing-list_003f" href="#Will-I-receive-spam-if-I-send-and_002for-subscribe-to-a-mailing-list_003f">5.5 Will I receive spam if I send and/or subscribe to a mailing list?</a></li>
<li><a id="toc-How-do-I-filter-mailing-list-messages_003f" href="#How-do-I-filter-mailing-list-messages_003f">5.6 How do I filter mailing list messages?</a></li>
<li><a id="toc-How-do-I-disable-mail-delivery-without-unsubscribing_003f-1" href="#How-do-I-disable-mail-delivery-without-unsubscribing_003f-1">5.7 How do I disable mail delivery without unsubscribing?</a></li>
<li><a id="toc-Why-is-the-mailing-list-munging-my-address_003f-1" href="#Why-is-the-mailing-list-munging-my-address_003f-1">5.8 Why is the mailing list munging my address?</a></li>
</ul></li>
<li><a id="toc-Rules-and-Etiquette" href="#Rules-and-Etiquette">6 Rules and Etiquette</a>
<ul class="no-bullet">
<li><a id="toc-What-are-the-rules-and-the-proper-etiquette_003f" href="#What-are-the-rules-and-the-proper-etiquette_003f">6.1 What are the rules and the proper etiquette?</a></li>
</ul></li>
<li><a id="toc-Help" href="#Help">7 Help</a>
<ul class="no-bullet">
<li><a id="toc-Why-am-I-not-receiving-any-messages_003f" href="#Why-am-I-not-receiving-any-messages_003f">7.1 Why am I not receiving any messages?</a></li>
<li><a id="toc-Why-are-my-sent-messages-not-showing-up_003f" href="#Why-are-my-sent-messages-not-showing-up_003f">7.2 Why are my sent messages not showing up?</a></li>
<li><a id="toc-Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f-1" href="#Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f-1">7.3 Why do I keep getting unsubscribed from ffmpeg-devel?</a></li>
<li><a id="toc-Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f-1" href="#Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f-1">7.4 Who do I contact if I have a problem with the mailing list?</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="General-Questions"></a>
<h2 class="chapter">1 General Questions<span class="pull-right"><a class="anchor hidden-xs" href="#General-Questions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-General-Questions" aria-hidden="true">TOC</a></span></h2>

<a name="What-is-a-mailing-list_003f"></a>
<h3 class="section">1.1 What is a mailing list?<span class="pull-right"><a class="anchor hidden-xs" href="#What-is-a-mailing-list_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-What-is-a-mailing-list_003f" aria-hidden="true">TOC</a></span></h3>

<p>A mailing list is not much different than emailing someone, but the
main difference is that your message is received by everyone who
subscribes to the list. It is somewhat like a forum but in email form.
</p>
<p>See the <a href="https://lists.ffmpeg.org/pipermail/ffmpeg-user/">ffmpeg-user archives</a>
for examples.
</p>
<a name="What-type-of-questions-can-I-ask_003f"></a>
<h3 class="section">1.2 What type of questions can I ask?<span class="pull-right"><a class="anchor hidden-xs" href="#What-type-of-questions-can-I-ask_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-What-type-of-questions-can-I-ask_003f" aria-hidden="true">TOC</a></span></h3>

<ul>
<li> <a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/">ffmpeg-user</a>:
For questions involving unscripted usage or compilation of the FFmpeg
command-line tools (<code>ffmpeg</code>, <code>ffprobe</code>, <code>ffplay</code>).

</li><li> <a href="https://lists.ffmpeg.org/mailman/listinfo/libav-user/">libav-user</a>:
For questions involving the FFmpeg libav* libraries (libavcodec,
libavformat, libavfilter, etc).

</li><li> <a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/">ffmpeg-devel</a>:
For discussions involving the development of FFmpeg and for submitting
patches. User questions should be asked at ffmpeg-user or libav-user.
</li></ul>

<p>To report a bug see <a href="https://ffmpeg.org/bugreports.html">https://ffmpeg.org/bugreports.html</a>.
</p>
<p>We cannot provide help for scripts and/or third-party tools.
</p>
<span id="How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f"></span><a name="How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f-1"></a>
<h3 class="section">1.3 How do I ask a question or send a message to a mailing list?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-ask-a-question-or-send-a-message-to-a-mailing-list_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>First you must <a href="#How-do-I-subscribe_003f">subscribe</a>. Then all you have to do is
send an email:
</p>
<ul>
<li> Email <a href="mailto:ffmpeg-user@ffmpeg.org">ffmpeg-user@ffmpeg.org</a> to send a message to the
ffmpeg-user mailing list.

</li><li> Email <a href="mailto:libav-user@ffmpeg.org">libav-user@ffmpeg.org</a> to send a message to the
libav-user mailing list.

</li><li> Email <a href="mailto:ffmpeg-devel@ffmpeg.org">ffmpeg-devel@ffmpeg.org</a> to send a message to the
ffmpeg-devel mailing list.
</li></ul>

<a name="Subscribing-_002f-Unsubscribing"></a>
<h2 class="chapter">2 Subscribing / Unsubscribing<span class="pull-right"><a class="anchor hidden-xs" href="#Subscribing-_002f-Unsubscribing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Subscribing-_002f-Unsubscribing" aria-hidden="true">TOC</a></span></h2>

<span id="How-do-I-subscribe_003f"></span><a name="How-do-I-subscribe_003f-1"></a>
<h3 class="section">2.1 How do I subscribe?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-subscribe_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-subscribe_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>Email <a href="mailto:ffmpeg-user-request@ffmpeg.org">ffmpeg-user-request@ffmpeg.org</a> with the subject
<em>subscribe</em>.
</p>
<p>Or visit the <a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/">ffmpeg-user mailing list info page</a>
and refer to the <em>Subscribing to ffmpeg-user</em> section.
</p>
<p>The process is the same for the other mailing lists.
</p>
<a name="How-do-I-unsubscribe_003f"></a>
<h3 class="section">2.2 How do I unsubscribe?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-unsubscribe_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-unsubscribe_003f" aria-hidden="true">TOC</a></span></h3>

<p>Email <a href="mailto:ffmpeg-user-request@ffmpeg.org">ffmpeg-user-request@ffmpeg.org</a> with subject <em>unsubscribe</em>.
</p>
<p>Or visit the <a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/">ffmpeg-user mailing list info page</a>,
scroll to bottom of page, enter your email address in the box, and click
the <em>Unsubscribe or edit options</em> button.
</p>
<p>The process is the same for the other mailing lists.
</p>
<p>Please avoid asking a mailing list admin to unsubscribe you unless you
are absolutely unable to do so by yourself. See <a href="#Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f">Who do I contact if I have a problem with the mailing list?</a>
</p>
<p>Note that it is possible to temporarily halt message delivery (vacation mode).
See <a href="#How-do-I-disable-mail-delivery-without-unsubscribing_003f">How do I disable mail delivery without unsubscribing?</a>
</p>
<a name="Moderation-Queue"></a>
<h2 class="chapter">3 Moderation Queue<span class="pull-right"><a class="anchor hidden-xs" href="#Moderation-Queue" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Moderation-Queue" aria-hidden="true">TOC</a></span></h2>
<span id="Why-is-my-message-awaiting-moderator-approval_003f"></span><a name="Why-is-my-message-awaiting-moderator-approval_003f-1"></a>
<h3 class="section">3.1 Why is my message awaiting moderator approval?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-my-message-awaiting-moderator-approval_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-my-message-awaiting-moderator-approval_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>Some messages are automatically held in the <em>moderation queue</em> and
must be manually approved by a mailing list admin:
</p>
<p>These are:
</p>
<ul>
<li> Messages that exceed the <a href="#What-is-the-message-size-limit_003f">message size limit</a>.

</li><li> Messages from users whose accounts have been set with the <em>moderation flag</em>
(very rarely occurs, but may if a user repeatedly ignores the rules
or is abusive towards others).
</li></ul>

<a name="How-long-does-it-take-for-my-message-in-the-moderation-queue-to-be-approved_003f"></a>
<h3 class="section">3.2 How long does it take for my message in the moderation queue to be approved?<span class="pull-right"><a class="anchor hidden-xs" href="#How-long-does-it-take-for-my-message-in-the-moderation-queue-to-be-approved_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-long-does-it-take-for-my-message-in-the-moderation-queue-to-be-approved_003f" aria-hidden="true">TOC</a></span></h3>

<p>The queue is not checked on a regular basis. You can ask on the
<tt>#ffmpeg-devel</tt> IRC channel on Libera Chat for someone to approve your message.
</p>
<span id="How-do-I-delete-my-message-in-the-moderation-queue_003f"></span><a name="How-do-I-delete-my-message-in-the-moderation-queue_003f-1"></a>
<h3 class="section">3.3 How do I delete my message in the moderation queue?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-delete-my-message-in-the-moderation-queue_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-delete-my-message-in-the-moderation-queue_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>You should have received an email with the subject <em>Your message to &lt;mailing list name&gt; awaits moderator approval</em>.
A link is in the message that will allow you to delete your message
unless a mailing list admin already approved or rejected it.
</p>
<a name="Archives"></a>
<h2 class="chapter">4 Archives<span class="pull-right"><a class="anchor hidden-xs" href="#Archives" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Archives" aria-hidden="true">TOC</a></span></h2>

<span id="Where-are-the-archives_003f"></span><a name="Where-are-the-archives_003f-1"></a>
<h3 class="section">4.1 Where are the archives?<span class="pull-right"><a class="anchor hidden-xs" href="#Where-are-the-archives_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Where-are-the-archives_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>See the <em>Archives</em> section on the <a href="https://ffmpeg.org/contact.html">FFmpeg Contact</a>
page for links to all FFmpeg mailing list archives.
</p>
<p>Note that the archives are split by month. Discussions that span
several months will be split into separate months in the archives.
</p>
<a name="How-do-I-reply-to-a-message-in-the-archives_003f"></a>
<h3 class="section">4.2 How do I reply to a message in the archives?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-reply-to-a-message-in-the-archives_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-reply-to-a-message-in-the-archives_003f" aria-hidden="true">TOC</a></span></h3>

<p>Click the email link at the top of the message just under the subject
title. The link will provide the proper headers to keep the message
within the thread.
</p>
<p>Note that you must be subscribed to send a message to the ffmpeg-user or
libav-user mailing lists.
</p>
<a name="How-do-I-search-the-archives_003f"></a>
<h3 class="section">4.3 How do I search the archives?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-search-the-archives_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-search-the-archives_003f" aria-hidden="true">TOC</a></span></h3>

<p>Perform a site search using your favorite search engine. Example:
</p>
<p><tt>site:lists.ffmpeg.org/pipermail/ffmpeg-user/ &quot;search term&quot;</tt>
</p>
<a name="Other"></a>
<h2 class="chapter">5 Other<span class="pull-right"><a class="anchor hidden-xs" href="#Other" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Other" aria-hidden="true">TOC</a></span></h2>

<a name="Is-there-an-alternative-to-the-mailing-list_003f"></a>
<h3 class="section">5.1 Is there an alternative to the mailing list?<span class="pull-right"><a class="anchor hidden-xs" href="#Is-there-an-alternative-to-the-mailing-list_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Is-there-an-alternative-to-the-mailing-list_003f" aria-hidden="true">TOC</a></span></h3>

<p>You can ask for help in the official <tt>#ffmpeg</tt> IRC channel on Libera Chat.
</p>
<p>Some users prefer the third-party <a href="http://www.ffmpeg-archive.org/">Nabble</a>
interface which presents the mailing lists in a typical forum layout.
</p>
<p>There are also numerous third-party help sites such as
<a href="https://superuser.com/tags/ffmpeg">Super User</a> and
<a href="https://www.reddit.com/r/ffmpeg/">r/ffmpeg on reddit</a>.
</p>
<span id="What-is-top_002dposting_003f"></span><a name="What-is-top_002dposting_003f-1"></a>
<h3 class="section">5.2 What is top-posting?<span class="pull-right"><a class="anchor hidden-xs" href="#What-is-top_002dposting_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-What-is-top_002dposting_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>See <a href="https://en.wikipedia.org/wiki/Posting_style#Top-posting">https://en.wikipedia.org/wiki/Posting_style#Top-posting</a>.
</p>
<p>Instead, use trimmed interleaved/inline replies (<a href="https://lists.ffmpeg.org/pipermail/ffmpeg-user/2017-April/035849.html">example</a>).
</p>
<span id="What-is-the-message-size-limit_003f"></span><a name="What-is-the-message-size-limit_003f-1"></a>
<h3 class="section">5.3 What is the message size limit?<span class="pull-right"><a class="anchor hidden-xs" href="#What-is-the-message-size-limit_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-What-is-the-message-size-limit_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>The message size limit is 1000 kilobytes. Please provide links to larger files
instead of attaching them.
</p>
<a name="Where-can-I-upload-sample-files_003f"></a>
<h3 class="section">5.4 Where can I upload sample files?<span class="pull-right"><a class="anchor hidden-xs" href="#Where-can-I-upload-sample-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Where-can-I-upload-sample-files_003f" aria-hidden="true">TOC</a></span></h3>

<p>Anywhere that is not too annoying for us to use.
</p>
<p>Google Drive and Dropbox are acceptable if you need a file host, and
<a href="https://0x0.st/">0x0.st</a> is good for files under 256 MiB.
</p>
<p>Small, short samples are preferred if possible.
</p>
<a name="Will-I-receive-spam-if-I-send-and_002for-subscribe-to-a-mailing-list_003f"></a>
<h3 class="section">5.5 Will I receive spam if I send and/or subscribe to a mailing list?<span class="pull-right"><a class="anchor hidden-xs" href="#Will-I-receive-spam-if-I-send-and_002for-subscribe-to-a-mailing-list_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Will-I-receive-spam-if-I-send-and_002for-subscribe-to-a-mailing-list_003f" aria-hidden="true">TOC</a></span></h3>

<p>Highly unlikely.
</p>
<ul>
<li> The list of subscribed users is not public.

</li><li> Email addresses in the archives are obfuscated.

</li><li> Several unique test email accounts were utilized and none have yet
received any spam.
</li></ul>

<p>However, you may see a spam in the mailing lists on rare occasions:
</p>
<ul>
<li> Spam in the moderation queue may be accidentally approved due to human
error.

</li><li> There have been a few messages from subscribed users who had their own
email addresses hacked and spam messages from (or appearing to be from)
the hacked account were sent to their contacts (a mailing list being a
contact in these cases).

</li><li> If you are subscribed to the bug tracker mailing list (ffmpeg-trac) you
may see the occasional spam as a false bug report, but we take measures
to try to prevent this.
</li></ul>

<a name="How-do-I-filter-mailing-list-messages_003f"></a>
<h3 class="section">5.6 How do I filter mailing list messages?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-filter-mailing-list-messages_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-filter-mailing-list-messages_003f" aria-hidden="true">TOC</a></span></h3>

<p>Use the <em>List-Id</em>. For example, the ffmpeg-user mailing list is
<tt>ffmpeg-user.ffmpeg.org</tt>. You can view the List-Id in the raw message
or headers.
</p>
<p>You can then filter the mailing list messages to their own folder.
</p>
<span id="How-do-I-disable-mail-delivery-without-unsubscribing_003f"></span><a name="How-do-I-disable-mail-delivery-without-unsubscribing_003f-1"></a>
<h3 class="section">5.7 How do I disable mail delivery without unsubscribing?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-disable-mail-delivery-without-unsubscribing_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-disable-mail-delivery-without-unsubscribing_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>Sometimes you may want to temporarily stop receiving all mailing list
messages. This &quot;vacation mode&quot; is simple to do:
</p>
<ol>
<li> Go to the <a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/">ffmpeg-user mailing list info page</a>

</li><li> Enter your email address in the box at very bottom of the page and click the
<em>Unsubscribe or edit options</em> box.

</li><li> Enter your password and click the <em>Log in</em> button.

</li><li> Look for the <em>Mail delivery</em> option. Here you can disable/enable mail
delivery. If you check <em>Set globally</em> it will apply your choice to all
other FFmpeg mailing lists you are subscribed to.
</li></ol>

<p>Alternatively, from your subscribed address, send a message to <a href="mailto:ffmpeg-user-request@ffmpeg.org">ffmpeg-user-request@ffmpeg.org</a>
with the subject <em>set delivery off</em>. To re-enable mail delivery send a
message to <a href="mailto:ffmpeg-user-request@ffmpeg.org">ffmpeg-user-request@ffmpeg.org</a> with the subject
<em>set delivery on</em>.
</p>
<span id="Why-is-the-mailing-list-munging-my-address_003f"></span><a name="Why-is-the-mailing-list-munging-my-address_003f-1"></a>
<h3 class="section">5.8 Why is the mailing list munging my address?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-the-mailing-list-munging-my-address_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-the-mailing-list-munging-my-address_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>This is due to subscribers that use an email service with a DMARC reject policy
which adds difficulties to mailing list operators.
</p>
<p>The mailing list must re-write (munge) the <em>From:</em> header for such users;
otherwise their email service will reject and bounce the message resulting in
automatic unsubscribing from the mailing list.
</p>
<p>When sending a message these users will see <em>via &lt;mailing list name&gt;</em>
added to their name and the <em>From:</em> address munged to the address of
the particular mailing list.
</p>
<p>If you want to avoid this then please use a different email service.
</p>
<p>Note that ffmpeg-devel does not apply any munging as it causes issues with
patch authorship. As a result users with an email service with a DMARC reject
policy may be automatically unsubscribed due to rejected and bounced messages.
</p>
<a name="Rules-and-Etiquette"></a>
<h2 class="chapter">6 Rules and Etiquette<span class="pull-right"><a class="anchor hidden-xs" href="#Rules-and-Etiquette" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Rules-and-Etiquette" aria-hidden="true">TOC</a></span></h2>

<a name="What-are-the-rules-and-the-proper-etiquette_003f"></a>
<h3 class="section">6.1 What are the rules and the proper etiquette?<span class="pull-right"><a class="anchor hidden-xs" href="#What-are-the-rules-and-the-proper-etiquette_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-What-are-the-rules-and-the-proper-etiquette_003f" aria-hidden="true">TOC</a></span></h3>

<p>There may seem to be many things to remember, but we want to help and
following these guidelines will allow you to get answers more quickly
and help avoid getting ignored.
</p>
<ul>
<li> Always show your actual, unscripted <code>ffmpeg</code> command and the
complete, uncut console output from your command.

</li><li> Use the most simple and minimal command that still shows the issue you
are encountering.

</li><li> Provide all necessary information so others can attempt to duplicate
your issue. This includes the actual command, complete uncut console
output, and any inputs that are required to duplicate the issue.

</li><li> Use the latest <code>ffmpeg</code> build you can get. See the <a href="https://ffmpeg.org/download.html">FFmpeg Download</a>
page for links to recent builds for Linux, macOS, and Windows. Or
compile from the current git master branch.

</li><li> Avoid <a href="https://en.wikipedia.org/wiki/Posting_style#Top-posting">top-posting</a>.
Also see <a href="#What-is-top_002dposting_003f">What is top-posting?</a>

</li><li> Avoid hijacking threads. Thread hijacking is replying to a message and
changing the subject line to something unrelated to the original thread.
Most email clients will still show the renamed message under the
original thread. This can be confusing and these types of messages are
often ignored.

</li><li> Do not send screenshots. Copy and paste console text instead of making
screenshots of the text.

</li><li> Avoid sending email disclaimers and legalese if possible as this is a
public list.

</li><li> Avoid using the <code>-loglevel debug</code>, <code>-loglevel quiet</code>, and
<code>-hide_banner</code> options unless requested to do so.

</li><li> If you attach files avoid compressing small files. Uncompressed is
preferred.

</li><li> Please do not send HTML-only messages. The mailing list will ignore the
HTML component of your message. Most mail clients will automatically
include a text component: this is what the mailing list will use.

</li><li> Configuring your mail client to break lines after 70 or so characters is
recommended.

</li><li> Avoid sending the same message to multiple mailing lists.

</li><li> Please follow our <a href="https://ffmpeg.org/community.html#Code-of-conduct">Code of Conduct</a>.
</li></ul>

<a name="Help"></a>
<h2 class="chapter">7 Help<span class="pull-right"><a class="anchor hidden-xs" href="#Help" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Help" aria-hidden="true">TOC</a></span></h2>

<a name="Why-am-I-not-receiving-any-messages_003f"></a>
<h3 class="section">7.1 Why am I not receiving any messages?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-am-I-not-receiving-any-messages_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-am-I-not-receiving-any-messages_003f" aria-hidden="true">TOC</a></span></h3>

<p>Some email providers have blacklists or spam filters that block or mark
the mailing list messages as false positives. Unfortunately, the user is
often not aware of this and is often out of their control.
</p>
<p>When possible we attempt to notify the provider to be removed from the
blacklists or filters.
</p>
<a name="Why-are-my-sent-messages-not-showing-up_003f"></a>
<h3 class="section">7.2 Why are my sent messages not showing up?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-are-my-sent-messages-not-showing-up_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-are-my-sent-messages-not-showing-up_003f" aria-hidden="true">TOC</a></span></h3>

<p>Excluding <a href="#Why-is-my-message-awaiting-moderator-approval_003f">messages that are held in the moderation queue</a>
there are a few other reasons why your messages may fail to appear:
</p>
<ul>
<li> HTML-only messages are ignored by the mailing lists. Most mail clients
automatically include a text component alongside HTML email: this is what
the mailing list will use. If it does not then consider your client to be
broken, because sending a text component along with the HTML component to
form a multi-part message is recommended by email standards.

</li><li> Check your spam folder.
</li></ul>

<span id="Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f"></span><a name="Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f-1"></a>
<h3 class="section">7.3 Why do I keep getting unsubscribed from ffmpeg-devel?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-do-I-keep-getting-unsubscribed-from-ffmpeg_002ddevel_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>Users with an email service that has a DMARC reject or quarantine policy may be
automatically unsubscribed from the ffmpeg-devel mailing list due to the mailing
list messages being continuously rejected and bounced back.
</p>
<p>Consider using a different email service.
</p>
<span id="Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f"></span><a name="Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f-1"></a>
<h3 class="section">7.4 Who do I contact if I have a problem with the mailing list?<span class="pull-right"><a class="anchor hidden-xs" href="#Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Who-do-I-contact-if-I-have-a-problem-with-the-mailing-list_003f-1" aria-hidden="true">TOC</a></span></h3>

<p>Send a message to <a href="mailto:ffmpeg-user-owner@ffmpeg.org">ffmpeg-user-owner@ffmpeg.org</a>.
</p>
<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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NUT
</title>
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Description" href="#Description">1 Description</a></li>
<li><a id="toc-Modes" href="#Modes">2 Modes</a>
<ul class="no-bullet">
<li><a id="toc-BROADCAST" href="#BROADCAST">2.1 BROADCAST</a></li>
<li><a id="toc-PIPE" href="#PIPE">2.2 PIPE</a></li>
</ul></li>
<li><a id="toc-Container_002dspecific-codec-tags" href="#Container_002dspecific-codec-tags">3 Container-specific codec tags</a>
<ul class="no-bullet">
<li><a id="toc-Generic-raw-YUVA-formats" href="#Generic-raw-YUVA-formats">3.1 Generic raw YUVA formats</a></li>
<li><a id="toc-Raw-Audio" href="#Raw-Audio">3.2 Raw Audio</a></li>
<li><a id="toc-Subtitles" href="#Subtitles">3.3 Subtitles</a></li>
<li><a id="toc-Raw-Data" href="#Raw-Data">3.4 Raw Data</a></li>
<li><a id="toc-Codecs" href="#Codecs">3.5 Codecs</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>NUT is a low overhead generic container format. It stores audio, video,
subtitle and user-defined streams in a simple, yet efficient, way.
</p>
<p>It was created by a group of FFmpeg and MPlayer developers in 2003
and was finalized in 2008.
</p>
<p>The official nut specification is at svn://svn.mplayerhq.hu/nut
In case of any differences between this text and the official specification,
the official specification shall prevail.
</p>
<a name="Modes"></a>
<h2 class="chapter">2 Modes<span class="pull-right"><a class="anchor hidden-xs" href="#Modes" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Modes" aria-hidden="true">TOC</a></span></h2>
<p>NUT has some variants signaled by using the flags field in its main header.
</p>
<table>
<tr><td width="40%">BROADCAST</td><td width="40%">Extend the syncpoint to report the sender wallclock</td></tr>
<tr><td width="40%">PIPE</td><td width="40%">Omit completely the syncpoint</td></tr>
</table>

<a name="BROADCAST"></a>
<h3 class="section">2.1 BROADCAST<span class="pull-right"><a class="anchor hidden-xs" href="#BROADCAST" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BROADCAST" aria-hidden="true">TOC</a></span></h3>

<p>The BROADCAST variant provides a secondary time reference to facilitate
detecting endpoint latency and network delays.
It assumes all the endpoint clocks are synchronized.
To be used in real-time scenarios.
</p>
<a name="PIPE"></a>
<h3 class="section">2.2 PIPE<span class="pull-right"><a class="anchor hidden-xs" href="#PIPE" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-PIPE" aria-hidden="true">TOC</a></span></h3>

<p>The PIPE variant assumes NUT is used as non-seekable intermediate container,
by not using syncpoint removes unneeded overhead and reduces the overall
memory usage.
</p>
<a name="Container_002dspecific-codec-tags"></a>
<h2 class="chapter">3 Container-specific codec tags<span class="pull-right"><a class="anchor hidden-xs" href="#Container_002dspecific-codec-tags" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Container_002dspecific-codec-tags" aria-hidden="true">TOC</a></span></h2>

<a name="Generic-raw-YUVA-formats"></a>
<h3 class="section">3.1 Generic raw YUVA formats<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-raw-YUVA-formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-raw-YUVA-formats" aria-hidden="true">TOC</a></span></h3>

<p>Since many exotic planar YUVA pixel formats are not considered by
the AVI/QuickTime FourCC lists, the following scheme is adopted for
representing them.
</p>
<p>The first two bytes can contain the values:
Y1 = only Y
Y2 = Y+A
Y3 = YUV
Y4 = YUVA
</p>
<p>The third byte represents the width and height chroma subsampling
values for the UV planes, that is the amount to shift the luma
width/height right to find the chroma width/height.
</p>
<p>The fourth byte is the number of bits used (8, 16, ...).
</p>
<p>If the order of bytes is inverted, that means that each component has
to be read big-endian.
</p>
<a name="Raw-Audio"></a>
<h3 class="section">3.2 Raw Audio<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Audio" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Audio" aria-hidden="true">TOC</a></span></h3>

<table>
<tr><td width="40%">ALAW</td><td width="40%">A-LAW</td></tr>
<tr><td width="40%">ULAW</td><td width="40%">MU-LAW</td></tr>
<tr><td width="40%">P&lt;type&gt;&lt;interleaving&gt;&lt;bits&gt;</td><td width="40%">little-endian PCM</td></tr>
<tr><td width="40%">&lt;bits&gt;&lt;interleaving&gt;&lt;type&gt;P</td><td width="40%">big-endian PCM</td></tr>
</table>

<p>&lt;type&gt; is S for signed integer, U for unsigned integer, F for IEEE float
&lt;interleaving&gt; is D for default, P is for planar.
&lt;bits&gt; is 8/16/24/32
</p>
<div class="example">
<pre class="example">PFD[32] would for example be signed 32 bit little-endian IEEE float
</pre></div>

<a name="Subtitles"></a>
<h3 class="section">3.3 Subtitles<span class="pull-right"><a class="anchor hidden-xs" href="#Subtitles" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Subtitles" aria-hidden="true">TOC</a></span></h3>

<table>
<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
<tr><td width="40%">SSA[0]</td><td width="40%">SubStation Alpha</td></tr>
<tr><td width="40%">DVDS</td><td width="40%">DVD subtitles</td></tr>
<tr><td width="40%">DVBS</td><td width="40%">DVB subtitles</td></tr>
</table>

<a name="Raw-Data"></a>
<h3 class="section">3.4 Raw Data<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Data" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Data" aria-hidden="true">TOC</a></span></h3>

<table>
<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
</table>

<a name="Codecs"></a>
<h3 class="section">3.5 Codecs<span class="pull-right"><a class="anchor hidden-xs" href="#Codecs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Codecs" aria-hidden="true">TOC</a></span></h3>

<table>
<tr><td width="40%">3IV1</td><td width="40%">non-compliant MPEG-4 generated by old 3ivx</td></tr>
<tr><td width="40%">ASV1</td><td width="40%">Asus Video</td></tr>
<tr><td width="40%">ASV2</td><td width="40%">Asus Video 2</td></tr>
<tr><td width="40%">CVID</td><td width="40%">Cinepak</td></tr>
<tr><td width="40%">CYUV</td><td width="40%">Creative YUV</td></tr>
<tr><td width="40%">DIVX</td><td width="40%">non-compliant MPEG-4 generated by old DivX</td></tr>
<tr><td width="40%">DUCK</td><td width="40%">Truemotion 1</td></tr>
<tr><td width="40%">FFV1</td><td width="40%">FFmpeg video 1</td></tr>
<tr><td width="40%">FFVH</td><td width="40%">FFmpeg Huffyuv</td></tr>
<tr><td width="40%">H261</td><td width="40%">ITU H.261</td></tr>
<tr><td width="40%">H262</td><td width="40%">ITU H.262</td></tr>
<tr><td width="40%">H263</td><td width="40%">ITU H.263</td></tr>
<tr><td width="40%">H264</td><td width="40%">ITU H.264</td></tr>
<tr><td width="40%">HFYU</td><td width="40%">Huffyuv</td></tr>
<tr><td width="40%">I263</td><td width="40%">Intel H.263</td></tr>
<tr><td width="40%">IV31</td><td width="40%">Indeo 3.1</td></tr>
<tr><td width="40%">IV32</td><td width="40%">Indeo 3.2</td></tr>
<tr><td width="40%">IV50</td><td width="40%">Indeo 5.0</td></tr>
<tr><td width="40%">LJPG</td><td width="40%">ITU JPEG (lossless)</td></tr>
<tr><td width="40%">MJLS</td><td width="40%">ITU JPEG-LS</td></tr>
<tr><td width="40%">MJPG</td><td width="40%">ITU JPEG</td></tr>
<tr><td width="40%">MPG4</td><td width="40%">MS MPEG-4v1 (not ISO MPEG-4)</td></tr>
<tr><td width="40%">MP42</td><td width="40%">MS MPEG-4v2</td></tr>
<tr><td width="40%">MP43</td><td width="40%">MS MPEG-4v3</td></tr>
<tr><td width="40%">MP4V</td><td width="40%">ISO MPEG-4 Part 2 Video (from old encoders)</td></tr>
<tr><td width="40%">mpg1</td><td width="40%">ISO MPEG-1 Video</td></tr>
<tr><td width="40%">mpg2</td><td width="40%">ISO MPEG-2 Video</td></tr>
<tr><td width="40%">MRLE</td><td width="40%">MS RLE</td></tr>
<tr><td width="40%">MSVC</td><td width="40%">MS Video 1</td></tr>
<tr><td width="40%">RT21</td><td width="40%">Indeo 2.1</td></tr>
<tr><td width="40%">RV10</td><td width="40%">RealVideo 1.0</td></tr>
<tr><td width="40%">RV20</td><td width="40%">RealVideo 2.0</td></tr>
<tr><td width="40%">RV30</td><td width="40%">RealVideo 3.0</td></tr>
<tr><td width="40%">RV40</td><td width="40%">RealVideo 4.0</td></tr>
<tr><td width="40%">SNOW</td><td width="40%">FFmpeg Snow</td></tr>
<tr><td width="40%">SVQ1</td><td width="40%">Sorenson Video 1</td></tr>
<tr><td width="40%">SVQ3</td><td width="40%">Sorenson Video 3</td></tr>
<tr><td width="40%">theo</td><td width="40%">Xiph Theora</td></tr>
<tr><td width="40%">TM20</td><td width="40%">Truemotion 2.0</td></tr>
<tr><td width="40%">UMP4</td><td width="40%">non-compliant MPEG-4 generated by UB Video MPEG-4</td></tr>
<tr><td width="40%">VCR1</td><td width="40%">ATI VCR1</td></tr>
<tr><td width="40%">VP30</td><td width="40%">VP 3.0</td></tr>
<tr><td width="40%">VP31</td><td width="40%">VP 3.1</td></tr>
<tr><td width="40%">VP50</td><td width="40%">VP 5.0</td></tr>
<tr><td width="40%">VP60</td><td width="40%">VP 6.0</td></tr>
<tr><td width="40%">VP61</td><td width="40%">VP 6.1</td></tr>
<tr><td width="40%">VP62</td><td width="40%">VP 6.2</td></tr>
<tr><td width="40%">VP70</td><td width="40%">VP 7.0</td></tr>
<tr><td width="40%">WMV1</td><td width="40%">MS WMV7</td></tr>
<tr><td width="40%">WMV2</td><td width="40%">MS WMV8</td></tr>
<tr><td width="40%">WMV3</td><td width="40%">MS WMV9</td></tr>
<tr><td width="40%">WV1F</td><td width="40%">non-compliant MPEG-4 generated by ?</td></tr>
<tr><td width="40%">WVC1</td><td width="40%">VC-1</td></tr>
<tr><td width="40%">XVID</td><td width="40%">non-compliant MPEG-4 generated by old Xvid</td></tr>
<tr><td width="40%">XVIX</td><td width="40%">non-compliant MPEG-4 generated by old Xvid with interlacing bug</td></tr>
</table>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<title>
Platform Specific Information
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
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Platform Specific Information
</h1>
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<a name="SEC_Top"></a>

<div class="Contents_element" id="SEC_Contents">
<h2 class="contents-heading">Table of Contents</h2>

<div class="contents">

<ul class="no-bullet">
<li><a id="toc-Unix_002dlike" href="#Unix_002dlike">1 Unix-like</a>
<ul class="no-bullet">
<li><a id="toc-Advanced-linking-configuration" href="#Advanced-linking-configuration">1.1 Advanced linking configuration</a></li>
<li><a id="toc-BSD" href="#BSD">1.2 BSD</a></li>
<li><a id="toc-_0028Open_0029Solaris" href="#g_t_0028Open_0029Solaris">1.3 (Open)Solaris</a></li>
<li><a id="toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029">1.4 Darwin (Mac OS X, iPhone)</a></li>
</ul></li>
<li><a id="toc-DOS" href="#DOS">2 DOS</a></li>
<li><a id="toc-OS_002f2" href="#OS_002f2">3 OS/2</a></li>
<li><a id="toc-Windows" href="#Windows">4 Windows</a>
<ul class="no-bullet">
<li><a id="toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64">4.1 Native Windows compilation using MinGW or MinGW-w64</a>
<ul class="no-bullet">
<li><a id="toc-Native-Windows-compilation-using-MSYS2" href="#Native-Windows-compilation-using-MSYS2">4.1.1 Native Windows compilation using MSYS2</a></li>
</ul></li>
<li><a id="toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows</a>
<ul class="no-bullet">
<li><a id="toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b">4.2.1 Linking to FFmpeg with Microsoft Visual C++</a></li>
</ul></li>
<li><a id="toc-Cross-compilation-for-Windows-with-Linux-1" href="#Cross-compilation-for-Windows-with-Linux-1">4.3 Cross compilation for Windows with Linux</a></li>
<li><a id="toc-Compilation-under-Cygwin" href="#Compilation-under-Cygwin">4.4 Compilation under Cygwin</a></li>
<li><a id="toc-Crosscompilation-for-Windows-under-Cygwin" href="#Crosscompilation-for-Windows-under-Cygwin">4.5 Crosscompilation for Windows under Cygwin</a></li>
</ul></li>
</ul>
</div>
</div>

<a name="Unix_002dlike"></a>
<h2 class="chapter">1 Unix-like<span class="pull-right"><a class="anchor hidden-xs" href="#Unix_002dlike" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Unix_002dlike" aria-hidden="true">TOC</a></span></h2>

<p>Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
</p>
<div class="example">
<pre class="example">$(gcc -print-prog-name=as) --version
</pre></div>

<p>If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass <code>--disable-asm</code>
to configure.
</p>
<a name="Advanced-linking-configuration"></a>
<h3 class="section">1.1 Advanced linking configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-linking-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-linking-configuration" aria-hidden="true">TOC</a></span></h3>

<p>If you compiled FFmpeg libraries statically and you want to use them to
build your own shared library, you may need to force PIC support (with
<code>--enable-pic</code> during FFmpeg configure) and add the following option
to your project LDFLAGS:
</p>
<div class="example">
<pre class="example">-Wl,-Bsymbolic
</pre></div>

<p>If your target platform requires position independent binaries, you should
pass the correct linking flag (e.g. <code>-pie</code>) to <code>--extra-ldexeflags</code>.
</p>
<a name="BSD"></a>
<h3 class="section">1.2 BSD<span class="pull-right"><a class="anchor hidden-xs" href="#BSD" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BSD" aria-hidden="true">TOC</a></span></h3>

<p>BSD make will not build FFmpeg, you need to install and use GNU Make
(<code>gmake</code>).
</p>
<a name="g_t_0028Open_0029Solaris"></a>
<h3 class="section">1.3 (Open)Solaris<span class="pull-right"><a class="anchor hidden-xs" href="#_0028Open_0029Solaris" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_0028Open_0029Solaris" aria-hidden="true">TOC</a></span></h3>

<p>GNU Make is required to build FFmpeg, so you have to invoke (<code>gmake</code>),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either <code>--extra-libs=/usr/lib/values-xpg6.o</code>
or <code>--extra-libs=/usr/lib/64/values-xpg6.o</code> to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
</p>
<div class="example">
<pre class="example">bash ./configure
</pre></div>

<span id="Darwin"></span><a name="Darwin-_0028Mac-OS-X_002c-iPhone_0029"></a>
<h3 class="section">1.4 Darwin (Mac OS X, iPhone)<span class="pull-right"><a class="anchor hidden-xs" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">TOC</a></span></h3>

<p>The toolchain provided with Xcode is sufficient to build the basic
unaccelerated code.
</p>
<p>Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
<a href="https://github.com/FFmpeg/gas-preprocessor">https://github.com/FFmpeg/gas-preprocessor</a> or
<a href="https://github.com/yuvi/gas-preprocessor">https://github.com/yuvi/gas-preprocessor</a>(currently outdated) to build the optimized
assembly functions. Put the Perl script somewhere
in your PATH, FFmpeg&rsquo;s configure will pick it up automatically.
</p>
<p>Mac OS X on amd64 and x86 requires <code>nasm</code> to build most of the
optimized assembly functions. <a href="http://www.finkproject.org/">Fink</a>,
<a href="https://wiki.gentoo.org/wiki/Project:Prefix">Gentoo Prefix</a>,
<a href="https://mxcl.github.com/homebrew/">Homebrew</a>
or <a href="http://www.macports.org">MacPorts</a> can easily provide it.
</p>

<a name="DOS"></a>
<h2 class="chapter">2 DOS<span class="pull-right"><a class="anchor hidden-xs" href="#DOS" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-DOS" aria-hidden="true">TOC</a></span></h2>

<p>Using a cross-compiler is preferred for various reasons.
<a href="http://www.delorie.com/howto/djgpp/linux-x-djgpp.html">http://www.delorie.com/howto/djgpp/linux-x-djgpp.html</a>
</p>

<a name="OS_002f2"></a>
<h2 class="chapter">3 OS/2<span class="pull-right"><a class="anchor hidden-xs" href="#OS_002f2" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OS_002f2" aria-hidden="true">TOC</a></span></h2>

<p>For information about compiling FFmpeg on OS/2 see
<a href="http://www.edm2.com/index.php/FFmpeg">http://www.edm2.com/index.php/FFmpeg</a>.
</p>

<a name="Windows"></a>
<h2 class="chapter">4 Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Windows" aria-hidden="true">TOC</a></span></h2>

<p>To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at <a href="http://ffmpeg.zeranoe.com/forum/">http://ffmpeg.zeranoe.com/forum/</a>.
</p>
<a name="Native-Windows-compilation-using-MinGW-or-MinGW_002dw64"></a>
<h3 class="section">4.1 Native Windows compilation using MinGW or MinGW-w64<span class="pull-right"><a class="anchor hidden-xs" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg can be built to run natively on Windows using the MinGW-w64
toolchain. Install the latest versions of MSYS2 and MinGW-w64 from
<a href="http://msys2.github.io/">http://msys2.github.io/</a> and/or <a href="http://mingw-w64.sourceforge.net/">http://mingw-w64.sourceforge.net/</a>.
You can find detailed installation instructions in the download section and
the FAQ.
</p>
<p>Notes:
</p>
<ul>
<li> Building for the MSYS environment is discouraged, MSYS2 provides a full
MinGW-w64 environment through <samp>mingw64_shell.bat</samp> or
<samp>mingw32_shell.bat</samp> that should be used instead of the environment
provided by <samp>msys2_shell.bat</samp>.

</li><li> Building using MSYS2 can be sped up by disabling implicit rules in the
Makefile by calling <code>make -r</code> instead of plain <code>make</code>. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example during
<code>make install</code>).

</li><li> In order to compile FFplay, you must have the MinGW development library
of <a href="http://www.libsdl.org/">SDL</a> and <code>pkg-config</code> installed.

</li><li> By using <code>./configure --enable-shared</code> when configuring FFmpeg,
you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
libavformat) as DLLs.

</li></ul>

<a name="Native-Windows-compilation-using-MSYS2"></a>
<h4 class="subsection">4.1.1 Native Windows compilation using MSYS2<span class="pull-right"><a class="anchor hidden-xs" href="#Native-Windows-compilation-using-MSYS2" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Native-Windows-compilation-using-MSYS2" aria-hidden="true">TOC</a></span></h4>

<p>The MSYS2 MinGW-w64 environment provides ready to use toolchains and dependencies
through <code>pacman</code>.
</p>
<p>Make sure to use <samp>mingw64_shell.bat</samp> or <samp>mingw32_shell.bat</samp> to have
the correct MinGW-w64 environment. The default install provides shortcuts to
them under <code>MinGW-w64 Win64 Shell</code> and <code>MinGW-w64 Win32 Shell</code>.
</p>
<div class="example">
<pre class="example"># normal msys2 packages
pacman -S make pkgconf diffutils

# mingw-w64 packages and toolchains
pacman -S mingw-w64-x86_64-nasm mingw-w64-x86_64-gcc mingw-w64-x86_64-SDL2
</pre></div>

<p>To target 32 bits replace <code>x86_64</code> with <code>i686</code> in the command above.
</p>
<a name="Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows"></a>
<h3 class="section">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">TOC</a></span></h3>

<p>FFmpeg can be built with MSVC 2013 or later.
</p>
<p>You will need the following prerequisites:
</p>
<ul>
<li> <a href="http://msys2.github.io/">MSYS2</a>
</li><li> <a href="http://www.nasm.us/">NASM</a>
(Also available via MSYS2&rsquo;s package manager.)
</li></ul>

<p>To set up a proper environment in MSYS2, you need to run <code>msys_shell.bat</code> from
the Visual Studio or Intel Compiler command prompt.
</p>
<p>Place <code>yasm.exe</code> somewhere in your <code>PATH</code>.
</p>
<p>Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the <code>LIB</code>
and <code>INCLUDE</code> environment variables to include the <strong>Windows-style</strong>
paths to these directories. Alternatively, you can try to use the
<code>--extra-cflags</code>/<code>--extra-ldflags</code> configure options.
</p>
<p>Finally, run:
</p>
<div class="example">
<pre class="example">For MSVC:
./configure --toolchain=msvc

For ICL:
./configure --toolchain=icl

make
make install
</pre></div>

<p>If you wish to compile shared libraries, add <code>--enable-shared</code> to your
configure options. Note that due to the way MSVC and ICL handle DLL imports and
exports, you cannot compile static and shared libraries at the same time, and
enabling shared libraries will automatically disable the static ones.
</p>
<p>Notes:
</p>
<ul>
<li> If you wish to build with zlib support, you will have to grab a compatible
zlib binary from somewhere, with an MSVC import lib, or if you wish to link
statically, you can follow the instructions below to build a compatible
<code>zlib.lib</code> with MSVC. Regardless of which method you use, you must still
follow step 3, or compilation will fail.
<ol>
<li> Grab the <a href="http://zlib.net/">zlib sources</a>.
</li><li> Edit <code>win32/Makefile.msc</code> so that it uses -MT instead of -MD, since
this is how FFmpeg is built as well.
</li><li> Edit <code>zconf.h</code> and remove its inclusion of <code>unistd.h</code>. This gets
erroneously included when building FFmpeg.
</li><li> Run <code>nmake -f win32/Makefile.msc</code>.
</li><li> Move <code>zlib.lib</code>, <code>zconf.h</code>, and <code>zlib.h</code> to somewhere MSVC
can see.
</li></ol>

</li><li> FFmpeg has been tested with the following on i686 and x86_64:
<ul>
<li> Visual Studio 2013 Pro and Express
</li><li> Intel Composer XE 2013
</li><li> Intel Composer XE 2013 SP1
</li></ul>
<p>Anything else is not officially supported.
</p>
</li></ul>

<a name="Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b"></a>
<h4 class="subsection">4.2.1 Linking to FFmpeg with Microsoft Visual C++<span class="pull-right"><a class="anchor hidden-xs" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">TOC</a></span></h4>

<p>If you plan to link with MSVC-built static libraries, you will need
to make sure you have <code>Runtime Library</code> set to
<code>Multi-threaded (/MT)</code> in your project&rsquo;s settings.
</p>
<p>You will need to define <code>inline</code> to something MSVC understands:
</p><div class="example">
<pre class="example">#define inline __inline
</pre></div>

<p>Also note, that as stated in <strong>Microsoft Visual C++</strong>, you will need
an MSVC-compatible <a href="http://code.google.com/p/msinttypes/">inttypes.h</a>.
</p>
<p>If you plan on using import libraries created by dlltool, you must
set <code>References</code> to <code>No (/OPT:NOREF)</code> under the linker optimization
settings, otherwise the resulting binaries will fail during runtime.
This is not required when using import libraries generated by <code>lib.exe</code>.
This issue is reported upstream at
<a href="http://sourceware.org/bugzilla/show_bug.cgi?id=12633">http://sourceware.org/bugzilla/show_bug.cgi?id=12633</a>.
</p>
<p>To create import libraries that work with the <code>/OPT:REF</code> option
(which is enabled by default in Release mode), follow these steps:
</p>
<ol>
<li> Open the <em>Visual Studio Command Prompt</em>.

<p>Alternatively, in a normal command line prompt, call <samp>vcvars32.bat</samp>
which sets up the environment variables for the Visual C++ tools
(the standard location for this file is something like
<samp>C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat</samp>).
</p>
</li><li> Enter the <samp>bin</samp> directory where the created LIB and DLL files
are stored.

</li><li> Generate new import libraries with <code>lib.exe</code>:

<div class="example">
<pre class="example">lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib
</pre></div>

<p>Replace <code>foo-version</code> and <code>foo</code> with the respective library names.
</p>
</li></ol>

<span id="Cross-compilation-for-Windows-with-Linux"></span><a name="Cross-compilation-for-Windows-with-Linux-1"></a>
<h3 class="section">4.3 Cross compilation for Windows with Linux<span class="pull-right"><a class="anchor hidden-xs" href="#Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">TOC</a></span></h3>

<p>You must use the MinGW cross compilation tools available at
<a href="http://www.mingw.org/">http://www.mingw.org/</a>.
</p>
<p>Then configure FFmpeg with the following options:
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
</pre></div>
<p>(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
</p>
<p>Then you can easily test FFmpeg with <a href="http://www.winehq.com/">Wine</a>.
</p>
<a name="Compilation-under-Cygwin"></a>
<h3 class="section">4.4 Compilation under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation-under-Cygwin" aria-hidden="true">TOC</a></span></h3>

<p>Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
</p>
<p>Install your Cygwin with all the &quot;Base&quot; packages, plus the
following &quot;Devel&quot; ones:
</p><div class="example">
<pre class="example">binutils, gcc4-core, make, git, mingw-runtime, texinfo
</pre></div>

<p>In order to run FATE you will also need the following &quot;Utils&quot; packages:
</p><div class="example">
<pre class="example">diffutils
</pre></div>

<p>If you want to build FFmpeg with additional libraries, download Cygwin
&quot;Devel&quot; packages for Ogg and Vorbis from any Cygwin packages repository:
</p><div class="example">
<pre class="example">libogg-devel, libvorbis-devel
</pre></div>

<p>These library packages are only available from
<a href="http://sourceware.org/cygwinports/">Cygwin Ports</a>:
</p>
<div class="example">
<pre class="example">yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
speex-devel, libtheora-devel, libxvidcore-devel
</pre></div>

<p>The recommendation for x264 is to build it from source, as it evolves too
quickly for Cygwin Ports to be up to date.
</p>
<a name="Crosscompilation-for-Windows-under-Cygwin"></a>
<h3 class="section">4.5 Crosscompilation for Windows under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">TOC</a></span></h3>

<p>With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
</p>
<p>Just install your Cygwin as explained before, plus these additional
&quot;Devel&quot; packages:
</p><div class="example">
<pre class="example">gcc-mingw-core, mingw-runtime, mingw-zlib
</pre></div>

<p>and add some special flags to your configure invocation.
</p>
<p>For a static build run
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
</pre></div>

<p>and for a build with shared libraries
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
</pre></div>

<p style="font-size: small;">
This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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/*
* AC-3 parser prototypes
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_AC3_PARSER_H
#define AVCODEC_AC3_PARSER_H

#include <stddef.h>
#include <stdint.h>

/**
* Extract the bitstream ID and the frame size from AC-3 data.
*/
int av_ac3_parse_header(const uint8_t *buf, size_t size,
uint8_t *bitstream_id, uint16_t *frame_size);


#endif /* AVCODEC_AC3_PARSER_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_ADTS_PARSER_H
#define AVCODEC_ADTS_PARSER_H

#include <stddef.h>
#include <stdint.h>

#define AV_AAC_ADTS_HEADER_SIZE 7

/**
* Extract the number of samples and frames from AAC data.
* @param[in] buf pointer to AAC data buffer
* @param[out] samples Pointer to where number of samples is written
* @param[out] frames Pointer to where number of frames is written
* @return Returns 0 on success, error code on failure.
*/
int av_adts_header_parse(const uint8_t *buf, uint32_t *samples,
uint8_t *frames);

#endif /* AVCODEC_ADTS_PARSER_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_AVDCT_H
#define AVCODEC_AVDCT_H

#include "libavutil/opt.h"

/**
* AVDCT context.
* @note function pointers can be NULL if the specific features have been
* disabled at build time.
*/
typedef struct AVDCT {
const AVClass *av_class;

void (*idct)(int16_t *block /* align 16 */);

/**
* IDCT input permutation.
* Several optimized IDCTs need a permutated input (relative to the
* normal order of the reference IDCT).
* This permutation must be performed before the idct_put/add.
* Note, normally this can be merged with the zigzag/alternate scan<br>
* An example to avoid confusion:
* - (->decode coeffs -> zigzag reorder -> dequant -> reference IDCT -> ...)
* - (x -> reference DCT -> reference IDCT -> x)
* - (x -> reference DCT -> simple_mmx_perm = idct_permutation
* -> simple_idct_mmx -> x)
* - (-> decode coeffs -> zigzag reorder -> simple_mmx_perm -> dequant
* -> simple_idct_mmx -> ...)
*/
uint8_t idct_permutation[64];

void (*fdct)(int16_t *block /* align 16 */);


/**
* DCT algorithm.
* must use AVOptions to set this field.
*/
int dct_algo;

/**
* IDCT algorithm.
* must use AVOptions to set this field.
*/
int idct_algo;

void (*get_pixels)(int16_t *block /* align 16 */,
const uint8_t *pixels /* align 8 */,
ptrdiff_t line_size);

int bits_per_sample;

void (*get_pixels_unaligned)(int16_t *block /* align 16 */,
const uint8_t *pixels,
ptrdiff_t line_size);
} AVDCT;

/**
* Allocates a AVDCT context.
* This needs to be initialized with avcodec_dct_init() after optionally
* configuring it with AVOptions.
*
* To free it use av_free()
*/
AVDCT *avcodec_dct_alloc(void);
int avcodec_dct_init(AVDCT *);

const AVClass *avcodec_dct_get_class(void);

#endif /* AVCODEC_AVDCT_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_AVFFT_H
#define AVCODEC_AVFFT_H

/**
* @file
* @ingroup lavc_fft
* FFT functions
*/

/**
* @defgroup lavc_fft FFT functions
* @ingroup lavc_misc
*
* @{
*/

typedef float FFTSample;

typedef struct FFTComplex {
FFTSample re, im;
} FFTComplex;

typedef struct FFTContext FFTContext;

/**
* Set up a complex FFT.
* @param nbits log2 of the length of the input array
* @param inverse if 0 perform the forward transform, if 1 perform the inverse
*/
FFTContext *av_fft_init(int nbits, int inverse);

/**
* Do the permutation needed BEFORE calling ff_fft_calc().
*/
void av_fft_permute(FFTContext *s, FFTComplex *z);

/**
* Do a complex FFT with the parameters defined in av_fft_init(). The
* input data must be permuted before. No 1.0/sqrt(n) normalization is done.
*/
void av_fft_calc(FFTContext *s, FFTComplex *z);

void av_fft_end(FFTContext *s);

FFTContext *av_mdct_init(int nbits, int inverse, double scale);
void av_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_mdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
void av_mdct_end(FFTContext *s);

/* Real Discrete Fourier Transform */

enum RDFTransformType {
DFT_R2C,
IDFT_C2R,
IDFT_R2C,
DFT_C2R,
};

typedef struct RDFTContext RDFTContext;

/**
* Set up a real FFT.
* @param nbits log2 of the length of the input array
* @param trans the type of transform
*/
RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans);
void av_rdft_calc(RDFTContext *s, FFTSample *data);
void av_rdft_end(RDFTContext *s);

/* Discrete Cosine Transform */

typedef struct DCTContext DCTContext;

enum DCTTransformType {
DCT_II = 0,
DCT_III,
DCT_I,
DST_I,
};

/**
* Set up DCT.
*
* @param nbits size of the input array:
* (1 << nbits) for DCT-II, DCT-III and DST-I
* (1 << nbits) + 1 for DCT-I
* @param type the type of transform
*
* @note the first element of the input of DST-I is ignored
*/
DCTContext *av_dct_init(int nbits, enum DCTTransformType type);
void av_dct_calc(DCTContext *s, FFTSample *data);
void av_dct_end (DCTContext *s);

/**
* @}
*/

#endif /* AVCODEC_AVFFT_H */

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/*
* Bitstream filters public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_BSF_H
#define AVCODEC_BSF_H

#include "libavutil/dict.h"
#include "libavutil/log.h"
#include "libavutil/rational.h"

#include "codec_id.h"
#include "codec_par.h"
#include "packet.h"

/**
* @defgroup lavc_bsf Bitstream filters
* @ingroup libavc
*
* Bitstream filters transform encoded media data without decoding it. This
* allows e.g. manipulating various header values. Bitstream filters operate on
* @ref AVPacket "AVPackets".
*
* The bitstream filtering API is centered around two structures:
* AVBitStreamFilter and AVBSFContext. The former represents a bitstream filter
* in abstract, the latter a specific filtering process. Obtain an
* AVBitStreamFilter using av_bsf_get_by_name() or av_bsf_iterate(), then pass
* it to av_bsf_alloc() to create an AVBSFContext. Fill in the user-settable
* AVBSFContext fields, as described in its documentation, then call
* av_bsf_init() to prepare the filter context for use.
*
* Submit packets for filtering using av_bsf_send_packet(), obtain filtered
* results with av_bsf_receive_packet(). When no more input packets will be
* sent, submit a NULL AVPacket to signal the end of the stream to the filter.
* av_bsf_receive_packet() will then return trailing packets, if any are
* produced by the filter.
*
* Finally, free the filter context with av_bsf_free().
* @{
*/

/**
* The bitstream filter state.
*
* This struct must be allocated with av_bsf_alloc() and freed with
* av_bsf_free().
*
* The fields in the struct will only be changed (by the caller or by the
* filter) as described in their documentation, and are to be considered
* immutable otherwise.
*/
typedef struct AVBSFContext {
/**
* A class for logging and AVOptions
*/
const AVClass *av_class;

/**
* The bitstream filter this context is an instance of.
*/
const struct AVBitStreamFilter *filter;

/**
* Opaque filter-specific private data. If filter->priv_class is non-NULL,
* this is an AVOptions-enabled struct.
*/
void *priv_data;

/**
* Parameters of the input stream. This field is allocated in
* av_bsf_alloc(), it needs to be filled by the caller before
* av_bsf_init().
*/
AVCodecParameters *par_in;

/**
* Parameters of the output stream. This field is allocated in
* av_bsf_alloc(), it is set by the filter in av_bsf_init().
*/
AVCodecParameters *par_out;

/**
* The timebase used for the timestamps of the input packets. Set by the
* caller before av_bsf_init().
*/
AVRational time_base_in;

/**
* The timebase used for the timestamps of the output packets. Set by the
* filter in av_bsf_init().
*/
AVRational time_base_out;
} AVBSFContext;

typedef struct AVBitStreamFilter {
const char *name;

/**
* A list of codec ids supported by the filter, terminated by
* AV_CODEC_ID_NONE.
* May be NULL, in that case the bitstream filter works with any codec id.
*/
const enum AVCodecID *codec_ids;

/**
* A class for the private data, used to declare bitstream filter private
* AVOptions. This field is NULL for bitstream filters that do not declare
* any options.
*
* If this field is non-NULL, the first member of the filter private data
* must be a pointer to AVClass, which will be set by libavcodec generic
* code to this class.
*/
const AVClass *priv_class;
} AVBitStreamFilter;

/**
* @return a bitstream filter with the specified name or NULL if no such
* bitstream filter exists.
*/
const AVBitStreamFilter *av_bsf_get_by_name(const char *name);

/**
* Iterate over all registered bitstream filters.
*
* @param opaque a pointer where libavcodec will store the iteration state. Must
* point to NULL to start the iteration.
*
* @return the next registered bitstream filter or NULL when the iteration is
* finished
*/
const AVBitStreamFilter *av_bsf_iterate(void **opaque);

/**
* Allocate a context for a given bitstream filter. The caller must fill in the
* context parameters as described in the documentation and then call
* av_bsf_init() before sending any data to the filter.
*
* @param filter the filter for which to allocate an instance.
* @param[out] ctx a pointer into which the pointer to the newly-allocated context
* will be written. It must be freed with av_bsf_free() after the
* filtering is done.
*
* @return 0 on success, a negative AVERROR code on failure
*/
int av_bsf_alloc(const AVBitStreamFilter *filter, AVBSFContext **ctx);

/**
* Prepare the filter for use, after all the parameters and options have been
* set.
*
* @param ctx a AVBSFContext previously allocated with av_bsf_alloc()
*/
int av_bsf_init(AVBSFContext *ctx);

/**
* Submit a packet for filtering.
*
* After sending each packet, the filter must be completely drained by calling
* av_bsf_receive_packet() repeatedly until it returns AVERROR(EAGAIN) or
* AVERROR_EOF.
*
* @param ctx an initialized AVBSFContext
* @param pkt the packet to filter. The bitstream filter will take ownership of
* the packet and reset the contents of pkt. pkt is not touched if an error occurs.
* If pkt is empty (i.e. NULL, or pkt->data is NULL and pkt->side_data_elems zero),
* it signals the end of the stream (i.e. no more non-empty packets will be sent;
* sending more empty packets does nothing) and will cause the filter to output
* any packets it may have buffered internally.
*
* @return
* - 0 on success.
* - AVERROR(EAGAIN) if packets need to be retrieved from the filter (using
* av_bsf_receive_packet()) before new input can be consumed.
* - Another negative AVERROR value if an error occurs.
*/
int av_bsf_send_packet(AVBSFContext *ctx, AVPacket *pkt);

/**
* Retrieve a filtered packet.
*
* @param ctx an initialized AVBSFContext
* @param[out] pkt this struct will be filled with the contents of the filtered
* packet. It is owned by the caller and must be freed using
* av_packet_unref() when it is no longer needed.
* This parameter should be "clean" (i.e. freshly allocated
* with av_packet_alloc() or unreffed with av_packet_unref())
* when this function is called. If this function returns
* successfully, the contents of pkt will be completely
* overwritten by the returned data. On failure, pkt is not
* touched.
*
* @return
* - 0 on success.
* - AVERROR(EAGAIN) if more packets need to be sent to the filter (using
* av_bsf_send_packet()) to get more output.
* - AVERROR_EOF if there will be no further output from the filter.
* - Another negative AVERROR value if an error occurs.
*
* @note one input packet may result in several output packets, so after sending
* a packet with av_bsf_send_packet(), this function needs to be called
* repeatedly until it stops returning 0. It is also possible for a filter to
* output fewer packets than were sent to it, so this function may return
* AVERROR(EAGAIN) immediately after a successful av_bsf_send_packet() call.
*/
int av_bsf_receive_packet(AVBSFContext *ctx, AVPacket *pkt);

/**
* Reset the internal bitstream filter state. Should be called e.g. when seeking.
*/
void av_bsf_flush(AVBSFContext *ctx);

/**
* Free a bitstream filter context and everything associated with it; write NULL
* into the supplied pointer.
*/
void av_bsf_free(AVBSFContext **ctx);

/**
* Get the AVClass for AVBSFContext. It can be used in combination with
* AV_OPT_SEARCH_FAKE_OBJ for examining options.
*
* @see av_opt_find().
*/
const AVClass *av_bsf_get_class(void);

/**
* Structure for chain/list of bitstream filters.
* Empty list can be allocated by av_bsf_list_alloc().
*/
typedef struct AVBSFList AVBSFList;

/**
* Allocate empty list of bitstream filters.
* The list must be later freed by av_bsf_list_free()
* or finalized by av_bsf_list_finalize().
*
* @return Pointer to @ref AVBSFList on success, NULL in case of failure
*/
AVBSFList *av_bsf_list_alloc(void);

/**
* Free list of bitstream filters.
*
* @param lst Pointer to pointer returned by av_bsf_list_alloc()
*/
void av_bsf_list_free(AVBSFList **lst);

/**
* Append bitstream filter to the list of bitstream filters.
*
* @param lst List to append to
* @param bsf Filter context to be appended
*
* @return >=0 on success, negative AVERROR in case of failure
*/
int av_bsf_list_append(AVBSFList *lst, AVBSFContext *bsf);

/**
* Construct new bitstream filter context given it's name and options
* and append it to the list of bitstream filters.
*
* @param lst List to append to
* @param bsf_name Name of the bitstream filter
* @param options Options for the bitstream filter, can be set to NULL
*
* @return >=0 on success, negative AVERROR in case of failure
*/
int av_bsf_list_append2(AVBSFList *lst, const char * bsf_name, AVDictionary **options);
/**
* Finalize list of bitstream filters.
*
* This function will transform @ref AVBSFList to single @ref AVBSFContext,
* so the whole chain of bitstream filters can be treated as single filter
* freshly allocated by av_bsf_alloc().
* If the call is successful, @ref AVBSFList structure is freed and lst
* will be set to NULL. In case of failure, caller is responsible for
* freeing the structure by av_bsf_list_free()
*
* @param lst Filter list structure to be transformed
* @param[out] bsf Pointer to be set to newly created @ref AVBSFContext structure
* representing the chain of bitstream filters
*
* @return >=0 on success, negative AVERROR in case of failure
*/
int av_bsf_list_finalize(AVBSFList **lst, AVBSFContext **bsf);

/**
* Parse string describing list of bitstream filters and create single
* @ref AVBSFContext describing the whole chain of bitstream filters.
* Resulting @ref AVBSFContext can be treated as any other @ref AVBSFContext freshly
* allocated by av_bsf_alloc().
*
* @param str String describing chain of bitstream filters in format
* `bsf1[=opt1=val1:opt2=val2][,bsf2]`
* @param[out] bsf Pointer to be set to newly created @ref AVBSFContext structure
* representing the chain of bitstream filters
*
* @return >=0 on success, negative AVERROR in case of failure
*/
int av_bsf_list_parse_str(const char *str, AVBSFContext **bsf);

/**
* Get null/pass-through bitstream filter.
*
* @param[out] bsf Pointer to be set to new instance of pass-through bitstream filter
*
* @return
*/
int av_bsf_get_null_filter(AVBSFContext **bsf);

/**
* @}
*/

#endif // AVCODEC_BSF_H

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ffmpeg/include/libavcodec/codec.h 파일 보기

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/*
* AVCodec public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_CODEC_H
#define AVCODEC_CODEC_H

#include <stdint.h>

#include "libavutil/avutil.h"
#include "libavutil/hwcontext.h"
#include "libavutil/log.h"
#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"
#include "libavutil/samplefmt.h"

#include "libavcodec/codec_id.h"
#include "libavcodec/version_major.h"

/**
* @addtogroup lavc_core
* @{
*/

/**
* Decoder can use draw_horiz_band callback.
*/
#define AV_CODEC_CAP_DRAW_HORIZ_BAND (1 << 0)
/**
* Codec uses get_buffer() or get_encode_buffer() for allocating buffers and
* supports custom allocators.
* If not set, it might not use get_buffer() or get_encode_buffer() at all, or
* use operations that assume the buffer was allocated by
* avcodec_default_get_buffer2 or avcodec_default_get_encode_buffer.
*/
#define AV_CODEC_CAP_DR1 (1 << 1)
/**
* Encoder or decoder requires flushing with NULL input at the end in order to
* give the complete and correct output.
*
* NOTE: If this flag is not set, the codec is guaranteed to never be fed with
* with NULL data. The user can still send NULL data to the public encode
* or decode function, but libavcodec will not pass it along to the codec
* unless this flag is set.
*
* Decoders:
* The decoder has a non-zero delay and needs to be fed with avpkt->data=NULL,
* avpkt->size=0 at the end to get the delayed data until the decoder no longer
* returns frames.
*
* Encoders:
* The encoder needs to be fed with NULL data at the end of encoding until the
* encoder no longer returns data.
*
* NOTE: For encoders implementing the AVCodec.encode2() function, setting this
* flag also means that the encoder must set the pts and duration for
* each output packet. If this flag is not set, the pts and duration will
* be determined by libavcodec from the input frame.
*/
#define AV_CODEC_CAP_DELAY (1 << 5)
/**
* Codec can be fed a final frame with a smaller size.
* This can be used to prevent truncation of the last audio samples.
*/
#define AV_CODEC_CAP_SMALL_LAST_FRAME (1 << 6)

/**
* Codec can output multiple frames per AVPacket
* Normally demuxers return one frame at a time, demuxers which do not do
* are connected to a parser to split what they return into proper frames.
* This flag is reserved to the very rare category of codecs which have a
* bitstream that cannot be split into frames without timeconsuming
* operations like full decoding. Demuxers carrying such bitstreams thus
* may return multiple frames in a packet. This has many disadvantages like
* prohibiting stream copy in many cases thus it should only be considered
* as a last resort.
*/
#define AV_CODEC_CAP_SUBFRAMES (1 << 8)
/**
* Codec is experimental and is thus avoided in favor of non experimental
* encoders
*/
#define AV_CODEC_CAP_EXPERIMENTAL (1 << 9)
/**
* Codec should fill in channel configuration and samplerate instead of container
*/
#define AV_CODEC_CAP_CHANNEL_CONF (1 << 10)
/**
* Codec supports frame-level multithreading.
*/
#define AV_CODEC_CAP_FRAME_THREADS (1 << 12)
/**
* Codec supports slice-based (or partition-based) multithreading.
*/
#define AV_CODEC_CAP_SLICE_THREADS (1 << 13)
/**
* Codec supports changed parameters at any point.
*/
#define AV_CODEC_CAP_PARAM_CHANGE (1 << 14)
/**
* Codec supports multithreading through a method other than slice- or
* frame-level multithreading. Typically this marks wrappers around
* multithreading-capable external libraries.
*/
#define AV_CODEC_CAP_OTHER_THREADS (1 << 15)
/**
* Audio encoder supports receiving a different number of samples in each call.
*/
#define AV_CODEC_CAP_VARIABLE_FRAME_SIZE (1 << 16)
/**
* Decoder is not a preferred choice for probing.
* This indicates that the decoder is not a good choice for probing.
* It could for example be an expensive to spin up hardware decoder,
* or it could simply not provide a lot of useful information about
* the stream.
* A decoder marked with this flag should only be used as last resort
* choice for probing.
*/
#define AV_CODEC_CAP_AVOID_PROBING (1 << 17)

/**
* Codec is backed by a hardware implementation. Typically used to
* identify a non-hwaccel hardware decoder. For information about hwaccels, use
* avcodec_get_hw_config() instead.
*/
#define AV_CODEC_CAP_HARDWARE (1 << 18)

/**
* Codec is potentially backed by a hardware implementation, but not
* necessarily. This is used instead of AV_CODEC_CAP_HARDWARE, if the
* implementation provides some sort of internal fallback.
*/
#define AV_CODEC_CAP_HYBRID (1 << 19)

/**
* This encoder can reorder user opaque values from input AVFrames and return
* them with corresponding output packets.
* @see AV_CODEC_FLAG_COPY_OPAQUE
*/
#define AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE (1 << 20)

/**
* This encoder can be flushed using avcodec_flush_buffers(). If this flag is
* not set, the encoder must be closed and reopened to ensure that no frames
* remain pending.
*/
#define AV_CODEC_CAP_ENCODER_FLUSH (1 << 21)

/**
* The encoder is able to output reconstructed frame data, i.e. raw frames that
* would be produced by decoding the encoded bitstream.
*
* Reconstructed frame output is enabled by the AV_CODEC_FLAG_RECON_FRAME flag.
*/
#define AV_CODEC_CAP_ENCODER_RECON_FRAME (1 << 22)

/**
* AVProfile.
*/
typedef struct AVProfile {
int profile;
const char *name; ///< short name for the profile
} AVProfile;

/**
* AVCodec.
*/
typedef struct AVCodec {
/**
* Name of the codec implementation.
* The name is globally unique among encoders and among decoders (but an
* encoder and a decoder can share the same name).
* This is the primary way to find a codec from the user perspective.
*/
const char *name;
/**
* Descriptive name for the codec, meant to be more human readable than name.
* You should use the NULL_IF_CONFIG_SMALL() macro to define it.
*/
const char *long_name;
enum AVMediaType type;
enum AVCodecID id;
/**
* Codec capabilities.
* see AV_CODEC_CAP_*
*/
int capabilities;
uint8_t max_lowres; ///< maximum value for lowres supported by the decoder
const AVRational *supported_framerates; ///< array of supported framerates, or NULL if any, array is terminated by {0,0}
const enum AVPixelFormat *pix_fmts; ///< array of supported pixel formats, or NULL if unknown, array is terminated by -1
const int *supported_samplerates; ///< array of supported audio samplerates, or NULL if unknown, array is terminated by 0
const enum AVSampleFormat *sample_fmts; ///< array of supported sample formats, or NULL if unknown, array is terminated by -1
#if FF_API_OLD_CHANNEL_LAYOUT
/**
* @deprecated use ch_layouts instead
*/
attribute_deprecated
const uint64_t *channel_layouts; ///< array of support channel layouts, or NULL if unknown. array is terminated by 0
#endif
const AVClass *priv_class; ///< AVClass for the private context
const AVProfile *profiles; ///< array of recognized profiles, or NULL if unknown, array is terminated by {FF_PROFILE_UNKNOWN}

/**
* Group name of the codec implementation.
* This is a short symbolic name of the wrapper backing this codec. A
* wrapper uses some kind of external implementation for the codec, such
* as an external library, or a codec implementation provided by the OS or
* the hardware.
* If this field is NULL, this is a builtin, libavcodec native codec.
* If non-NULL, this will be the suffix in AVCodec.name in most cases
* (usually AVCodec.name will be of the form "<codec_name>_<wrapper_name>").
*/
const char *wrapper_name;

/**
* Array of supported channel layouts, terminated with a zeroed layout.
*/
const AVChannelLayout *ch_layouts;
} AVCodec;

/**
* Iterate over all registered codecs.
*
* @param opaque a pointer where libavcodec will store the iteration state. Must
* point to NULL to start the iteration.
*
* @return the next registered codec or NULL when the iteration is
* finished
*/
const AVCodec *av_codec_iterate(void **opaque);

/**
* Find a registered decoder with a matching codec ID.
*
* @param id AVCodecID of the requested decoder
* @return A decoder if one was found, NULL otherwise.
*/
const AVCodec *avcodec_find_decoder(enum AVCodecID id);

/**
* Find a registered decoder with the specified name.
*
* @param name name of the requested decoder
* @return A decoder if one was found, NULL otherwise.
*/
const AVCodec *avcodec_find_decoder_by_name(const char *name);

/**
* Find a registered encoder with a matching codec ID.
*
* @param id AVCodecID of the requested encoder
* @return An encoder if one was found, NULL otherwise.
*/
const AVCodec *avcodec_find_encoder(enum AVCodecID id);

/**
* Find a registered encoder with the specified name.
*
* @param name name of the requested encoder
* @return An encoder if one was found, NULL otherwise.
*/
const AVCodec *avcodec_find_encoder_by_name(const char *name);
/**
* @return a non-zero number if codec is an encoder, zero otherwise
*/
int av_codec_is_encoder(const AVCodec *codec);

/**
* @return a non-zero number if codec is a decoder, zero otherwise
*/
int av_codec_is_decoder(const AVCodec *codec);

/**
* Return a name for the specified profile, if available.
*
* @param codec the codec that is searched for the given profile
* @param profile the profile value for which a name is requested
* @return A name for the profile if found, NULL otherwise.
*/
const char *av_get_profile_name(const AVCodec *codec, int profile);

enum {
/**
* The codec supports this format via the hw_device_ctx interface.
*
* When selecting this format, AVCodecContext.hw_device_ctx should
* have been set to a device of the specified type before calling
* avcodec_open2().
*/
AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX = 0x01,
/**
* The codec supports this format via the hw_frames_ctx interface.
*
* When selecting this format for a decoder,
* AVCodecContext.hw_frames_ctx should be set to a suitable frames
* context inside the get_format() callback. The frames context
* must have been created on a device of the specified type.
*
* When selecting this format for an encoder,
* AVCodecContext.hw_frames_ctx should be set to the context which
* will be used for the input frames before calling avcodec_open2().
*/
AV_CODEC_HW_CONFIG_METHOD_HW_FRAMES_CTX = 0x02,
/**
* The codec supports this format by some internal method.
*
* This format can be selected without any additional configuration -
* no device or frames context is required.
*/
AV_CODEC_HW_CONFIG_METHOD_INTERNAL = 0x04,
/**
* The codec supports this format by some ad-hoc method.
*
* Additional settings and/or function calls are required. See the
* codec-specific documentation for details. (Methods requiring
* this sort of configuration are deprecated and others should be
* used in preference.)
*/
AV_CODEC_HW_CONFIG_METHOD_AD_HOC = 0x08,
};

typedef struct AVCodecHWConfig {
/**
* For decoders, a hardware pixel format which that decoder may be
* able to decode to if suitable hardware is available.
*
* For encoders, a pixel format which the encoder may be able to
* accept. If set to AV_PIX_FMT_NONE, this applies to all pixel
* formats supported by the codec.
*/
enum AVPixelFormat pix_fmt;
/**
* Bit set of AV_CODEC_HW_CONFIG_METHOD_* flags, describing the possible
* setup methods which can be used with this configuration.
*/
int methods;
/**
* The device type associated with the configuration.
*
* Must be set for AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX and
* AV_CODEC_HW_CONFIG_METHOD_HW_FRAMES_CTX, otherwise unused.
*/
enum AVHWDeviceType device_type;
} AVCodecHWConfig;

/**
* Retrieve supported hardware configurations for a codec.
*
* Values of index from zero to some maximum return the indexed configuration
* descriptor; all other values return NULL. If the codec does not support
* any hardware configurations then it will always return NULL.
*/
const AVCodecHWConfig *avcodec_get_hw_config(const AVCodec *codec, int index);

/**
* @}
*/

#endif /* AVCODEC_CODEC_H */

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ffmpeg/include/libavcodec/codec_desc.h 파일 보기

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/*
* Codec descriptors public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_CODEC_DESC_H
#define AVCODEC_CODEC_DESC_H

#include "libavutil/avutil.h"

#include "codec_id.h"

/**
* @addtogroup lavc_core
* @{
*/

/**
* This struct describes the properties of a single codec described by an
* AVCodecID.
* @see avcodec_descriptor_get()
*/
typedef struct AVCodecDescriptor {
enum AVCodecID id;
enum AVMediaType type;
/**
* Name of the codec described by this descriptor. It is non-empty and
* unique for each codec descriptor. It should contain alphanumeric
* characters and '_' only.
*/
const char *name;
/**
* A more descriptive name for this codec. May be NULL.
*/
const char *long_name;
/**
* Codec properties, a combination of AV_CODEC_PROP_* flags.
*/
int props;
/**
* MIME type(s) associated with the codec.
* May be NULL; if not, a NULL-terminated array of MIME types.
* The first item is always non-NULL and is the preferred MIME type.
*/
const char *const *mime_types;
/**
* If non-NULL, an array of profiles recognized for this codec.
* Terminated with FF_PROFILE_UNKNOWN.
*/
const struct AVProfile *profiles;
} AVCodecDescriptor;

/**
* Codec uses only intra compression.
* Video and audio codecs only.
*/
#define AV_CODEC_PROP_INTRA_ONLY (1 << 0)
/**
* Codec supports lossy compression. Audio and video codecs only.
* @note a codec may support both lossy and lossless
* compression modes
*/
#define AV_CODEC_PROP_LOSSY (1 << 1)
/**
* Codec supports lossless compression. Audio and video codecs only.
*/
#define AV_CODEC_PROP_LOSSLESS (1 << 2)
/**
* Codec supports frame reordering. That is, the coded order (the order in which
* the encoded packets are output by the encoders / stored / input to the
* decoders) may be different from the presentation order of the corresponding
* frames.
*
* For codecs that do not have this property set, PTS and DTS should always be
* equal.
*/
#define AV_CODEC_PROP_REORDER (1 << 3)
/**
* Subtitle codec is bitmap based
* Decoded AVSubtitle data can be read from the AVSubtitleRect->pict field.
*/
#define AV_CODEC_PROP_BITMAP_SUB (1 << 16)
/**
* Subtitle codec is text based.
* Decoded AVSubtitle data can be read from the AVSubtitleRect->ass field.
*/
#define AV_CODEC_PROP_TEXT_SUB (1 << 17)

/**
* @return descriptor for given codec ID or NULL if no descriptor exists.
*/
const AVCodecDescriptor *avcodec_descriptor_get(enum AVCodecID id);

/**
* Iterate over all codec descriptors known to libavcodec.
*
* @param prev previous descriptor. NULL to get the first descriptor.
*
* @return next descriptor or NULL after the last descriptor
*/
const AVCodecDescriptor *avcodec_descriptor_next(const AVCodecDescriptor *prev);

/**
* @return codec descriptor with the given name or NULL if no such descriptor
* exists.
*/
const AVCodecDescriptor *avcodec_descriptor_get_by_name(const char *name);

/**
* @}
*/

#endif // AVCODEC_CODEC_DESC_H

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ffmpeg/include/libavcodec/codec_id.h 파일 보기

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/*
* Codec IDs
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_CODEC_ID_H
#define AVCODEC_CODEC_ID_H

#include "libavutil/avutil.h"
#include "libavutil/samplefmt.h"

#include "version_major.h"

/**
* @addtogroup lavc_core
* @{
*/

/**
* Identify the syntax and semantics of the bitstream.
* The principle is roughly:
* Two decoders with the same ID can decode the same streams.
* Two encoders with the same ID can encode compatible streams.
* There may be slight deviations from the principle due to implementation
* details.
*
* If you add a codec ID to this list, add it so that
* 1. no value of an existing codec ID changes (that would break ABI),
* 2. it is as close as possible to similar codecs
*
* After adding new codec IDs, do not forget to add an entry to the codec
* descriptor list and bump libavcodec minor version.
*/
enum AVCodecID {
AV_CODEC_ID_NONE,

/* video codecs */
AV_CODEC_ID_MPEG1VIDEO,
AV_CODEC_ID_MPEG2VIDEO, ///< preferred ID for MPEG-1/2 video decoding
AV_CODEC_ID_H261,
AV_CODEC_ID_H263,
AV_CODEC_ID_RV10,
AV_CODEC_ID_RV20,
AV_CODEC_ID_MJPEG,
AV_CODEC_ID_MJPEGB,
AV_CODEC_ID_LJPEG,
AV_CODEC_ID_SP5X,
AV_CODEC_ID_JPEGLS,
AV_CODEC_ID_MPEG4,
AV_CODEC_ID_RAWVIDEO,
AV_CODEC_ID_MSMPEG4V1,
AV_CODEC_ID_MSMPEG4V2,
AV_CODEC_ID_MSMPEG4V3,
AV_CODEC_ID_WMV1,
AV_CODEC_ID_WMV2,
AV_CODEC_ID_H263P,
AV_CODEC_ID_H263I,
AV_CODEC_ID_FLV1,
AV_CODEC_ID_SVQ1,
AV_CODEC_ID_SVQ3,
AV_CODEC_ID_DVVIDEO,
AV_CODEC_ID_HUFFYUV,
AV_CODEC_ID_CYUV,
AV_CODEC_ID_H264,
AV_CODEC_ID_INDEO3,
AV_CODEC_ID_VP3,
AV_CODEC_ID_THEORA,
AV_CODEC_ID_ASV1,
AV_CODEC_ID_ASV2,
AV_CODEC_ID_FFV1,
AV_CODEC_ID_4XM,
AV_CODEC_ID_VCR1,
AV_CODEC_ID_CLJR,
AV_CODEC_ID_MDEC,
AV_CODEC_ID_ROQ,
AV_CODEC_ID_INTERPLAY_VIDEO,
AV_CODEC_ID_XAN_WC3,
AV_CODEC_ID_XAN_WC4,
AV_CODEC_ID_RPZA,
AV_CODEC_ID_CINEPAK,
AV_CODEC_ID_WS_VQA,
AV_CODEC_ID_MSRLE,
AV_CODEC_ID_MSVIDEO1,
AV_CODEC_ID_IDCIN,
AV_CODEC_ID_8BPS,
AV_CODEC_ID_SMC,
AV_CODEC_ID_FLIC,
AV_CODEC_ID_TRUEMOTION1,
AV_CODEC_ID_VMDVIDEO,
AV_CODEC_ID_MSZH,
AV_CODEC_ID_ZLIB,
AV_CODEC_ID_QTRLE,
AV_CODEC_ID_TSCC,
AV_CODEC_ID_ULTI,
AV_CODEC_ID_QDRAW,
AV_CODEC_ID_VIXL,
AV_CODEC_ID_QPEG,
AV_CODEC_ID_PNG,
AV_CODEC_ID_PPM,
AV_CODEC_ID_PBM,
AV_CODEC_ID_PGM,
AV_CODEC_ID_PGMYUV,
AV_CODEC_ID_PAM,
AV_CODEC_ID_FFVHUFF,
AV_CODEC_ID_RV30,
AV_CODEC_ID_RV40,
AV_CODEC_ID_VC1,
AV_CODEC_ID_WMV3,
AV_CODEC_ID_LOCO,
AV_CODEC_ID_WNV1,
AV_CODEC_ID_AASC,
AV_CODEC_ID_INDEO2,
AV_CODEC_ID_FRAPS,
AV_CODEC_ID_TRUEMOTION2,
AV_CODEC_ID_BMP,
AV_CODEC_ID_CSCD,
AV_CODEC_ID_MMVIDEO,
AV_CODEC_ID_ZMBV,
AV_CODEC_ID_AVS,
AV_CODEC_ID_SMACKVIDEO,
AV_CODEC_ID_NUV,
AV_CODEC_ID_KMVC,
AV_CODEC_ID_FLASHSV,
AV_CODEC_ID_CAVS,
AV_CODEC_ID_JPEG2000,
AV_CODEC_ID_VMNC,
AV_CODEC_ID_VP5,
AV_CODEC_ID_VP6,
AV_CODEC_ID_VP6F,
AV_CODEC_ID_TARGA,
AV_CODEC_ID_DSICINVIDEO,
AV_CODEC_ID_TIERTEXSEQVIDEO,
AV_CODEC_ID_TIFF,
AV_CODEC_ID_GIF,
AV_CODEC_ID_DXA,
AV_CODEC_ID_DNXHD,
AV_CODEC_ID_THP,
AV_CODEC_ID_SGI,
AV_CODEC_ID_C93,
AV_CODEC_ID_BETHSOFTVID,
AV_CODEC_ID_PTX,
AV_CODEC_ID_TXD,
AV_CODEC_ID_VP6A,
AV_CODEC_ID_AMV,
AV_CODEC_ID_VB,
AV_CODEC_ID_PCX,
AV_CODEC_ID_SUNRAST,
AV_CODEC_ID_INDEO4,
AV_CODEC_ID_INDEO5,
AV_CODEC_ID_MIMIC,
AV_CODEC_ID_RL2,
AV_CODEC_ID_ESCAPE124,
AV_CODEC_ID_DIRAC,
AV_CODEC_ID_BFI,
AV_CODEC_ID_CMV,
AV_CODEC_ID_MOTIONPIXELS,
AV_CODEC_ID_TGV,
AV_CODEC_ID_TGQ,
AV_CODEC_ID_TQI,
AV_CODEC_ID_AURA,
AV_CODEC_ID_AURA2,
AV_CODEC_ID_V210X,
AV_CODEC_ID_TMV,
AV_CODEC_ID_V210,
AV_CODEC_ID_DPX,
AV_CODEC_ID_MAD,
AV_CODEC_ID_FRWU,
AV_CODEC_ID_FLASHSV2,
AV_CODEC_ID_CDGRAPHICS,
AV_CODEC_ID_R210,
AV_CODEC_ID_ANM,
AV_CODEC_ID_BINKVIDEO,
AV_CODEC_ID_IFF_ILBM,
#define AV_CODEC_ID_IFF_BYTERUN1 AV_CODEC_ID_IFF_ILBM
AV_CODEC_ID_KGV1,
AV_CODEC_ID_YOP,
AV_CODEC_ID_VP8,
AV_CODEC_ID_PICTOR,
AV_CODEC_ID_ANSI,
AV_CODEC_ID_A64_MULTI,
AV_CODEC_ID_A64_MULTI5,
AV_CODEC_ID_R10K,
AV_CODEC_ID_MXPEG,
AV_CODEC_ID_LAGARITH,
AV_CODEC_ID_PRORES,
AV_CODEC_ID_JV,
AV_CODEC_ID_DFA,
AV_CODEC_ID_WMV3IMAGE,
AV_CODEC_ID_VC1IMAGE,
AV_CODEC_ID_UTVIDEO,
AV_CODEC_ID_BMV_VIDEO,
AV_CODEC_ID_VBLE,
AV_CODEC_ID_DXTORY,
AV_CODEC_ID_V410,
AV_CODEC_ID_XWD,
AV_CODEC_ID_CDXL,
AV_CODEC_ID_XBM,
AV_CODEC_ID_ZEROCODEC,
AV_CODEC_ID_MSS1,
AV_CODEC_ID_MSA1,
AV_CODEC_ID_TSCC2,
AV_CODEC_ID_MTS2,
AV_CODEC_ID_CLLC,
AV_CODEC_ID_MSS2,
AV_CODEC_ID_VP9,
AV_CODEC_ID_AIC,
AV_CODEC_ID_ESCAPE130,
AV_CODEC_ID_G2M,
AV_CODEC_ID_WEBP,
AV_CODEC_ID_HNM4_VIDEO,
AV_CODEC_ID_HEVC,
#define AV_CODEC_ID_H265 AV_CODEC_ID_HEVC
AV_CODEC_ID_FIC,
AV_CODEC_ID_ALIAS_PIX,
AV_CODEC_ID_BRENDER_PIX,
AV_CODEC_ID_PAF_VIDEO,
AV_CODEC_ID_EXR,
AV_CODEC_ID_VP7,
AV_CODEC_ID_SANM,
AV_CODEC_ID_SGIRLE,
AV_CODEC_ID_MVC1,
AV_CODEC_ID_MVC2,
AV_CODEC_ID_HQX,
AV_CODEC_ID_TDSC,
AV_CODEC_ID_HQ_HQA,
AV_CODEC_ID_HAP,
AV_CODEC_ID_DDS,
AV_CODEC_ID_DXV,
AV_CODEC_ID_SCREENPRESSO,
AV_CODEC_ID_RSCC,
AV_CODEC_ID_AVS2,
AV_CODEC_ID_PGX,
AV_CODEC_ID_AVS3,
AV_CODEC_ID_MSP2,
AV_CODEC_ID_VVC,
#define AV_CODEC_ID_H266 AV_CODEC_ID_VVC
AV_CODEC_ID_Y41P,
AV_CODEC_ID_AVRP,
AV_CODEC_ID_012V,
AV_CODEC_ID_AVUI,
#if FF_API_AYUV_CODECID
AV_CODEC_ID_AYUV,
#endif
AV_CODEC_ID_TARGA_Y216,
AV_CODEC_ID_V308,
AV_CODEC_ID_V408,
AV_CODEC_ID_YUV4,
AV_CODEC_ID_AVRN,
AV_CODEC_ID_CPIA,
AV_CODEC_ID_XFACE,
AV_CODEC_ID_SNOW,
AV_CODEC_ID_SMVJPEG,
AV_CODEC_ID_APNG,
AV_CODEC_ID_DAALA,
AV_CODEC_ID_CFHD,
AV_CODEC_ID_TRUEMOTION2RT,
AV_CODEC_ID_M101,
AV_CODEC_ID_MAGICYUV,
AV_CODEC_ID_SHEERVIDEO,
AV_CODEC_ID_YLC,
AV_CODEC_ID_PSD,
AV_CODEC_ID_PIXLET,
AV_CODEC_ID_SPEEDHQ,
AV_CODEC_ID_FMVC,
AV_CODEC_ID_SCPR,
AV_CODEC_ID_CLEARVIDEO,
AV_CODEC_ID_XPM,
AV_CODEC_ID_AV1,
AV_CODEC_ID_BITPACKED,
AV_CODEC_ID_MSCC,
AV_CODEC_ID_SRGC,
AV_CODEC_ID_SVG,
AV_CODEC_ID_GDV,
AV_CODEC_ID_FITS,
AV_CODEC_ID_IMM4,
AV_CODEC_ID_PROSUMER,
AV_CODEC_ID_MWSC,
AV_CODEC_ID_WCMV,
AV_CODEC_ID_RASC,
AV_CODEC_ID_HYMT,
AV_CODEC_ID_ARBC,
AV_CODEC_ID_AGM,
AV_CODEC_ID_LSCR,
AV_CODEC_ID_VP4,
AV_CODEC_ID_IMM5,
AV_CODEC_ID_MVDV,
AV_CODEC_ID_MVHA,
AV_CODEC_ID_CDTOONS,
AV_CODEC_ID_MV30,
AV_CODEC_ID_NOTCHLC,
AV_CODEC_ID_PFM,
AV_CODEC_ID_MOBICLIP,
AV_CODEC_ID_PHOTOCD,
AV_CODEC_ID_IPU,
AV_CODEC_ID_ARGO,
AV_CODEC_ID_CRI,
AV_CODEC_ID_SIMBIOSIS_IMX,
AV_CODEC_ID_SGA_VIDEO,
AV_CODEC_ID_GEM,
AV_CODEC_ID_VBN,
AV_CODEC_ID_JPEGXL,
AV_CODEC_ID_QOI,
AV_CODEC_ID_PHM,
AV_CODEC_ID_RADIANCE_HDR,
AV_CODEC_ID_WBMP,
AV_CODEC_ID_MEDIA100,
AV_CODEC_ID_VQC,

/* various PCM "codecs" */
AV_CODEC_ID_FIRST_AUDIO = 0x10000, ///< A dummy id pointing at the start of audio codecs
AV_CODEC_ID_PCM_S16LE = 0x10000,
AV_CODEC_ID_PCM_S16BE,
AV_CODEC_ID_PCM_U16LE,
AV_CODEC_ID_PCM_U16BE,
AV_CODEC_ID_PCM_S8,
AV_CODEC_ID_PCM_U8,
AV_CODEC_ID_PCM_MULAW,
AV_CODEC_ID_PCM_ALAW,
AV_CODEC_ID_PCM_S32LE,
AV_CODEC_ID_PCM_S32BE,
AV_CODEC_ID_PCM_U32LE,
AV_CODEC_ID_PCM_U32BE,
AV_CODEC_ID_PCM_S24LE,
AV_CODEC_ID_PCM_S24BE,
AV_CODEC_ID_PCM_U24LE,
AV_CODEC_ID_PCM_U24BE,
AV_CODEC_ID_PCM_S24DAUD,
AV_CODEC_ID_PCM_ZORK,
AV_CODEC_ID_PCM_S16LE_PLANAR,
AV_CODEC_ID_PCM_DVD,
AV_CODEC_ID_PCM_F32BE,
AV_CODEC_ID_PCM_F32LE,
AV_CODEC_ID_PCM_F64BE,
AV_CODEC_ID_PCM_F64LE,
AV_CODEC_ID_PCM_BLURAY,
AV_CODEC_ID_PCM_LXF,
AV_CODEC_ID_S302M,
AV_CODEC_ID_PCM_S8_PLANAR,
AV_CODEC_ID_PCM_S24LE_PLANAR,
AV_CODEC_ID_PCM_S32LE_PLANAR,
AV_CODEC_ID_PCM_S16BE_PLANAR,
AV_CODEC_ID_PCM_S64LE,
AV_CODEC_ID_PCM_S64BE,
AV_CODEC_ID_PCM_F16LE,
AV_CODEC_ID_PCM_F24LE,
AV_CODEC_ID_PCM_VIDC,
AV_CODEC_ID_PCM_SGA,

/* various ADPCM codecs */
AV_CODEC_ID_ADPCM_IMA_QT = 0x11000,
AV_CODEC_ID_ADPCM_IMA_WAV,
AV_CODEC_ID_ADPCM_IMA_DK3,
AV_CODEC_ID_ADPCM_IMA_DK4,
AV_CODEC_ID_ADPCM_IMA_WS,
AV_CODEC_ID_ADPCM_IMA_SMJPEG,
AV_CODEC_ID_ADPCM_MS,
AV_CODEC_ID_ADPCM_4XM,
AV_CODEC_ID_ADPCM_XA,
AV_CODEC_ID_ADPCM_ADX,
AV_CODEC_ID_ADPCM_EA,
AV_CODEC_ID_ADPCM_G726,
AV_CODEC_ID_ADPCM_CT,
AV_CODEC_ID_ADPCM_SWF,
AV_CODEC_ID_ADPCM_YAMAHA,
AV_CODEC_ID_ADPCM_SBPRO_4,
AV_CODEC_ID_ADPCM_SBPRO_3,
AV_CODEC_ID_ADPCM_SBPRO_2,
AV_CODEC_ID_ADPCM_THP,
AV_CODEC_ID_ADPCM_IMA_AMV,
AV_CODEC_ID_ADPCM_EA_R1,
AV_CODEC_ID_ADPCM_EA_R3,
AV_CODEC_ID_ADPCM_EA_R2,
AV_CODEC_ID_ADPCM_IMA_EA_SEAD,
AV_CODEC_ID_ADPCM_IMA_EA_EACS,
AV_CODEC_ID_ADPCM_EA_XAS,
AV_CODEC_ID_ADPCM_EA_MAXIS_XA,
AV_CODEC_ID_ADPCM_IMA_ISS,
AV_CODEC_ID_ADPCM_G722,
AV_CODEC_ID_ADPCM_IMA_APC,
AV_CODEC_ID_ADPCM_VIMA,
AV_CODEC_ID_ADPCM_AFC,
AV_CODEC_ID_ADPCM_IMA_OKI,
AV_CODEC_ID_ADPCM_DTK,
AV_CODEC_ID_ADPCM_IMA_RAD,
AV_CODEC_ID_ADPCM_G726LE,
AV_CODEC_ID_ADPCM_THP_LE,
AV_CODEC_ID_ADPCM_PSX,
AV_CODEC_ID_ADPCM_AICA,
AV_CODEC_ID_ADPCM_IMA_DAT4,
AV_CODEC_ID_ADPCM_MTAF,
AV_CODEC_ID_ADPCM_AGM,
AV_CODEC_ID_ADPCM_ARGO,
AV_CODEC_ID_ADPCM_IMA_SSI,
AV_CODEC_ID_ADPCM_ZORK,
AV_CODEC_ID_ADPCM_IMA_APM,
AV_CODEC_ID_ADPCM_IMA_ALP,
AV_CODEC_ID_ADPCM_IMA_MTF,
AV_CODEC_ID_ADPCM_IMA_CUNNING,
AV_CODEC_ID_ADPCM_IMA_MOFLEX,
AV_CODEC_ID_ADPCM_IMA_ACORN,
AV_CODEC_ID_ADPCM_XMD,

/* AMR */
AV_CODEC_ID_AMR_NB = 0x12000,
AV_CODEC_ID_AMR_WB,

/* RealAudio codecs*/
AV_CODEC_ID_RA_144 = 0x13000,
AV_CODEC_ID_RA_288,

/* various DPCM codecs */
AV_CODEC_ID_ROQ_DPCM = 0x14000,
AV_CODEC_ID_INTERPLAY_DPCM,
AV_CODEC_ID_XAN_DPCM,
AV_CODEC_ID_SOL_DPCM,
AV_CODEC_ID_SDX2_DPCM,
AV_CODEC_ID_GREMLIN_DPCM,
AV_CODEC_ID_DERF_DPCM,
AV_CODEC_ID_WADY_DPCM,
AV_CODEC_ID_CBD2_DPCM,

/* audio codecs */
AV_CODEC_ID_MP2 = 0x15000,
AV_CODEC_ID_MP3, ///< preferred ID for decoding MPEG audio layer 1, 2 or 3
AV_CODEC_ID_AAC,
AV_CODEC_ID_AC3,
AV_CODEC_ID_DTS,
AV_CODEC_ID_VORBIS,
AV_CODEC_ID_DVAUDIO,
AV_CODEC_ID_WMAV1,
AV_CODEC_ID_WMAV2,
AV_CODEC_ID_MACE3,
AV_CODEC_ID_MACE6,
AV_CODEC_ID_VMDAUDIO,
AV_CODEC_ID_FLAC,
AV_CODEC_ID_MP3ADU,
AV_CODEC_ID_MP3ON4,
AV_CODEC_ID_SHORTEN,
AV_CODEC_ID_ALAC,
AV_CODEC_ID_WESTWOOD_SND1,
AV_CODEC_ID_GSM, ///< as in Berlin toast format
AV_CODEC_ID_QDM2,
AV_CODEC_ID_COOK,
AV_CODEC_ID_TRUESPEECH,
AV_CODEC_ID_TTA,
AV_CODEC_ID_SMACKAUDIO,
AV_CODEC_ID_QCELP,
AV_CODEC_ID_WAVPACK,
AV_CODEC_ID_DSICINAUDIO,
AV_CODEC_ID_IMC,
AV_CODEC_ID_MUSEPACK7,
AV_CODEC_ID_MLP,
AV_CODEC_ID_GSM_MS, /* as found in WAV */
AV_CODEC_ID_ATRAC3,
AV_CODEC_ID_APE,
AV_CODEC_ID_NELLYMOSER,
AV_CODEC_ID_MUSEPACK8,
AV_CODEC_ID_SPEEX,
AV_CODEC_ID_WMAVOICE,
AV_CODEC_ID_WMAPRO,
AV_CODEC_ID_WMALOSSLESS,
AV_CODEC_ID_ATRAC3P,
AV_CODEC_ID_EAC3,
AV_CODEC_ID_SIPR,
AV_CODEC_ID_MP1,
AV_CODEC_ID_TWINVQ,
AV_CODEC_ID_TRUEHD,
AV_CODEC_ID_MP4ALS,
AV_CODEC_ID_ATRAC1,
AV_CODEC_ID_BINKAUDIO_RDFT,
AV_CODEC_ID_BINKAUDIO_DCT,
AV_CODEC_ID_AAC_LATM,
AV_CODEC_ID_QDMC,
AV_CODEC_ID_CELT,
AV_CODEC_ID_G723_1,
AV_CODEC_ID_G729,
AV_CODEC_ID_8SVX_EXP,
AV_CODEC_ID_8SVX_FIB,
AV_CODEC_ID_BMV_AUDIO,
AV_CODEC_ID_RALF,
AV_CODEC_ID_IAC,
AV_CODEC_ID_ILBC,
AV_CODEC_ID_OPUS,
AV_CODEC_ID_COMFORT_NOISE,
AV_CODEC_ID_TAK,
AV_CODEC_ID_METASOUND,
AV_CODEC_ID_PAF_AUDIO,
AV_CODEC_ID_ON2AVC,
AV_CODEC_ID_DSS_SP,
AV_CODEC_ID_CODEC2,
AV_CODEC_ID_FFWAVESYNTH,
AV_CODEC_ID_SONIC,
AV_CODEC_ID_SONIC_LS,
AV_CODEC_ID_EVRC,
AV_CODEC_ID_SMV,
AV_CODEC_ID_DSD_LSBF,
AV_CODEC_ID_DSD_MSBF,
AV_CODEC_ID_DSD_LSBF_PLANAR,
AV_CODEC_ID_DSD_MSBF_PLANAR,
AV_CODEC_ID_4GV,
AV_CODEC_ID_INTERPLAY_ACM,
AV_CODEC_ID_XMA1,
AV_CODEC_ID_XMA2,
AV_CODEC_ID_DST,
AV_CODEC_ID_ATRAC3AL,
AV_CODEC_ID_ATRAC3PAL,
AV_CODEC_ID_DOLBY_E,
AV_CODEC_ID_APTX,
AV_CODEC_ID_APTX_HD,
AV_CODEC_ID_SBC,
AV_CODEC_ID_ATRAC9,
AV_CODEC_ID_HCOM,
AV_CODEC_ID_ACELP_KELVIN,
AV_CODEC_ID_MPEGH_3D_AUDIO,
AV_CODEC_ID_SIREN,
AV_CODEC_ID_HCA,
AV_CODEC_ID_FASTAUDIO,
AV_CODEC_ID_MSNSIREN,
AV_CODEC_ID_DFPWM,
AV_CODEC_ID_BONK,
AV_CODEC_ID_MISC4,
AV_CODEC_ID_APAC,
AV_CODEC_ID_FTR,
AV_CODEC_ID_WAVARC,
AV_CODEC_ID_RKA,

/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
AV_CODEC_ID_DVD_SUBTITLE = 0x17000,
AV_CODEC_ID_DVB_SUBTITLE,
AV_CODEC_ID_TEXT, ///< raw UTF-8 text
AV_CODEC_ID_XSUB,
AV_CODEC_ID_SSA,
AV_CODEC_ID_MOV_TEXT,
AV_CODEC_ID_HDMV_PGS_SUBTITLE,
AV_CODEC_ID_DVB_TELETEXT,
AV_CODEC_ID_SRT,
AV_CODEC_ID_MICRODVD,
AV_CODEC_ID_EIA_608,
AV_CODEC_ID_JACOSUB,
AV_CODEC_ID_SAMI,
AV_CODEC_ID_REALTEXT,
AV_CODEC_ID_STL,
AV_CODEC_ID_SUBVIEWER1,
AV_CODEC_ID_SUBVIEWER,
AV_CODEC_ID_SUBRIP,
AV_CODEC_ID_WEBVTT,
AV_CODEC_ID_MPL2,
AV_CODEC_ID_VPLAYER,
AV_CODEC_ID_PJS,
AV_CODEC_ID_ASS,
AV_CODEC_ID_HDMV_TEXT_SUBTITLE,
AV_CODEC_ID_TTML,
AV_CODEC_ID_ARIB_CAPTION,

/* other specific kind of codecs (generally used for attachments) */
AV_CODEC_ID_FIRST_UNKNOWN = 0x18000, ///< A dummy ID pointing at the start of various fake codecs.
AV_CODEC_ID_TTF = 0x18000,

AV_CODEC_ID_SCTE_35, ///< Contain timestamp estimated through PCR of program stream.
AV_CODEC_ID_EPG,
AV_CODEC_ID_BINTEXT,
AV_CODEC_ID_XBIN,
AV_CODEC_ID_IDF,
AV_CODEC_ID_OTF,
AV_CODEC_ID_SMPTE_KLV,
AV_CODEC_ID_DVD_NAV,
AV_CODEC_ID_TIMED_ID3,
AV_CODEC_ID_BIN_DATA,


AV_CODEC_ID_PROBE = 0x19000, ///< codec_id is not known (like AV_CODEC_ID_NONE) but lavf should attempt to identify it

AV_CODEC_ID_MPEG2TS = 0x20000, /**< _FAKE_ codec to indicate a raw MPEG-2 TS
* stream (only used by libavformat) */
AV_CODEC_ID_MPEG4SYSTEMS = 0x20001, /**< _FAKE_ codec to indicate a MPEG-4 Systems
* stream (only used by libavformat) */
AV_CODEC_ID_FFMETADATA = 0x21000, ///< Dummy codec for streams containing only metadata information.
AV_CODEC_ID_WRAPPED_AVFRAME = 0x21001, ///< Passthrough codec, AVFrames wrapped in AVPacket
/**
* Dummy null video codec, useful mainly for development and debugging.
* Null encoder/decoder discard all input and never return any output.
*/
AV_CODEC_ID_VNULL,
/**
* Dummy null audio codec, useful mainly for development and debugging.
* Null encoder/decoder discard all input and never return any output.
*/
AV_CODEC_ID_ANULL,
};

/**
* Get the type of the given codec.
*/
enum AVMediaType avcodec_get_type(enum AVCodecID codec_id);

/**
* Get the name of a codec.
* @return a static string identifying the codec; never NULL
*/
const char *avcodec_get_name(enum AVCodecID id);

/**
* Return codec bits per sample.
*
* @param[in] codec_id the codec
* @return Number of bits per sample or zero if unknown for the given codec.
*/
int av_get_bits_per_sample(enum AVCodecID codec_id);

/**
* Return codec bits per sample.
* Only return non-zero if the bits per sample is exactly correct, not an
* approximation.
*
* @param[in] codec_id the codec
* @return Number of bits per sample or zero if unknown for the given codec.
*/
int av_get_exact_bits_per_sample(enum AVCodecID codec_id);

/**
* Return a name for the specified profile, if available.
*
* @param codec_id the ID of the codec to which the requested profile belongs
* @param profile the profile value for which a name is requested
* @return A name for the profile if found, NULL otherwise.
*
* @note unlike av_get_profile_name(), which searches a list of profiles
* supported by a specific decoder or encoder implementation, this
* function searches the list of profiles from the AVCodecDescriptor
*/
const char *avcodec_profile_name(enum AVCodecID codec_id, int profile);

/**
* Return the PCM codec associated with a sample format.
* @param be endianness, 0 for little, 1 for big,
* -1 (or anything else) for native
* @return AV_CODEC_ID_PCM_* or AV_CODEC_ID_NONE
*/
enum AVCodecID av_get_pcm_codec(enum AVSampleFormat fmt, int be);

/**
* @}
*/

#endif // AVCODEC_CODEC_ID_H

+ 247
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ffmpeg/include/libavcodec/codec_par.h 파일 보기

@@ -0,0 +1,247 @@
/*
* Codec parameters public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_CODEC_PAR_H
#define AVCODEC_CODEC_PAR_H

#include <stdint.h>

#include "libavutil/avutil.h"
#include "libavutil/channel_layout.h"
#include "libavutil/rational.h"
#include "libavutil/pixfmt.h"

#include "codec_id.h"

/**
* @addtogroup lavc_core
* @{
*/

enum AVFieldOrder {
AV_FIELD_UNKNOWN,
AV_FIELD_PROGRESSIVE,
AV_FIELD_TT, ///< Top coded_first, top displayed first
AV_FIELD_BB, ///< Bottom coded first, bottom displayed first
AV_FIELD_TB, ///< Top coded first, bottom displayed first
AV_FIELD_BT, ///< Bottom coded first, top displayed first
};

/**
* This struct describes the properties of an encoded stream.
*
* sizeof(AVCodecParameters) is not a part of the public ABI, this struct must
* be allocated with avcodec_parameters_alloc() and freed with
* avcodec_parameters_free().
*/
typedef struct AVCodecParameters {
/**
* General type of the encoded data.
*/
enum AVMediaType codec_type;
/**
* Specific type of the encoded data (the codec used).
*/
enum AVCodecID codec_id;
/**
* Additional information about the codec (corresponds to the AVI FOURCC).
*/
uint32_t codec_tag;

/**
* Extra binary data needed for initializing the decoder, codec-dependent.
*
* Must be allocated with av_malloc() and will be freed by
* avcodec_parameters_free(). The allocated size of extradata must be at
* least extradata_size + AV_INPUT_BUFFER_PADDING_SIZE, with the padding
* bytes zeroed.
*/
uint8_t *extradata;
/**
* Size of the extradata content in bytes.
*/
int extradata_size;

/**
* - video: the pixel format, the value corresponds to enum AVPixelFormat.
* - audio: the sample format, the value corresponds to enum AVSampleFormat.
*/
int format;

/**
* The average bitrate of the encoded data (in bits per second).
*/
int64_t bit_rate;

/**
* The number of bits per sample in the codedwords.
*
* This is basically the bitrate per sample. It is mandatory for a bunch of
* formats to actually decode them. It's the number of bits for one sample in
* the actual coded bitstream.
*
* This could be for example 4 for ADPCM
* For PCM formats this matches bits_per_raw_sample
* Can be 0
*/
int bits_per_coded_sample;

/**
* This is the number of valid bits in each output sample. If the
* sample format has more bits, the least significant bits are additional
* padding bits, which are always 0. Use right shifts to reduce the sample
* to its actual size. For example, audio formats with 24 bit samples will
* have bits_per_raw_sample set to 24, and format set to AV_SAMPLE_FMT_S32.
* To get the original sample use "(int32_t)sample >> 8"."
*
* For ADPCM this might be 12 or 16 or similar
* Can be 0
*/
int bits_per_raw_sample;

/**
* Codec-specific bitstream restrictions that the stream conforms to.
*/
int profile;
int level;

/**
* Video only. The dimensions of the video frame in pixels.
*/
int width;
int height;

/**
* Video only. The aspect ratio (width / height) which a single pixel
* should have when displayed.
*
* When the aspect ratio is unknown / undefined, the numerator should be
* set to 0 (the denominator may have any value).
*/
AVRational sample_aspect_ratio;

/**
* Video only. The order of the fields in interlaced video.
*/
enum AVFieldOrder field_order;

/**
* Video only. Additional colorspace characteristics.
*/
enum AVColorRange color_range;
enum AVColorPrimaries color_primaries;
enum AVColorTransferCharacteristic color_trc;
enum AVColorSpace color_space;
enum AVChromaLocation chroma_location;

/**
* Video only. Number of delayed frames.
*/
int video_delay;

#if FF_API_OLD_CHANNEL_LAYOUT
/**
* Audio only. The channel layout bitmask. May be 0 if the channel layout is
* unknown or unspecified, otherwise the number of bits set must be equal to
* the channels field.
* @deprecated use ch_layout
*/
attribute_deprecated
uint64_t channel_layout;
/**
* Audio only. The number of audio channels.
* @deprecated use ch_layout.nb_channels
*/
attribute_deprecated
int channels;
#endif
/**
* Audio only. The number of audio samples per second.
*/
int sample_rate;
/**
* Audio only. The number of bytes per coded audio frame, required by some
* formats.
*
* Corresponds to nBlockAlign in WAVEFORMATEX.
*/
int block_align;
/**
* Audio only. Audio frame size, if known. Required by some formats to be static.
*/
int frame_size;

/**
* Audio only. The amount of padding (in samples) inserted by the encoder at
* the beginning of the audio. I.e. this number of leading decoded samples
* must be discarded by the caller to get the original audio without leading
* padding.
*/
int initial_padding;
/**
* Audio only. The amount of padding (in samples) appended by the encoder to
* the end of the audio. I.e. this number of decoded samples must be
* discarded by the caller from the end of the stream to get the original
* audio without any trailing padding.
*/
int trailing_padding;
/**
* Audio only. Number of samples to skip after a discontinuity.
*/
int seek_preroll;

/**
* Audio only. The channel layout and number of channels.
*/
AVChannelLayout ch_layout;
} AVCodecParameters;

/**
* Allocate a new AVCodecParameters and set its fields to default values
* (unknown/invalid/0). The returned struct must be freed with
* avcodec_parameters_free().
*/
AVCodecParameters *avcodec_parameters_alloc(void);

/**
* Free an AVCodecParameters instance and everything associated with it and
* write NULL to the supplied pointer.
*/
void avcodec_parameters_free(AVCodecParameters **par);

/**
* Copy the contents of src to dst. Any allocated fields in dst are freed and
* replaced with newly allocated duplicates of the corresponding fields in src.
*
* @return >= 0 on success, a negative AVERROR code on failure.
*/
int avcodec_parameters_copy(AVCodecParameters *dst, const AVCodecParameters *src);

/**
* This function is the same as av_get_audio_frame_duration(), except it works
* with AVCodecParameters instead of an AVCodecContext.
*/
int av_get_audio_frame_duration2(AVCodecParameters *par, int frame_bytes);

/**
* @}
*/

#endif // AVCODEC_CODEC_PAR_H

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ffmpeg/include/libavcodec/d3d11va.h 파일 보기

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/*
* Direct3D11 HW acceleration
*
* copyright (c) 2009 Laurent Aimar
* copyright (c) 2015 Steve Lhomme
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_D3D11VA_H
#define AVCODEC_D3D11VA_H

/**
* @file
* @ingroup lavc_codec_hwaccel_d3d11va
* Public libavcodec D3D11VA header.
*/

#if !defined(_WIN32_WINNT) || _WIN32_WINNT < 0x0602
#undef _WIN32_WINNT
#define _WIN32_WINNT 0x0602
#endif

#include <stdint.h>
#include <d3d11.h>

/**
* @defgroup lavc_codec_hwaccel_d3d11va Direct3D11
* @ingroup lavc_codec_hwaccel
*
* @{
*/

#define FF_DXVA2_WORKAROUND_SCALING_LIST_ZIGZAG 1 ///< Work around for Direct3D11 and old UVD/UVD+ ATI video cards
#define FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO 2 ///< Work around for Direct3D11 and old Intel GPUs with ClearVideo interface

/**
* This structure is used to provides the necessary configurations and data
* to the Direct3D11 FFmpeg HWAccel implementation.
*
* The application must make it available as AVCodecContext.hwaccel_context.
*
* Use av_d3d11va_alloc_context() exclusively to allocate an AVD3D11VAContext.
*/
typedef struct AVD3D11VAContext {
/**
* D3D11 decoder object
*/
ID3D11VideoDecoder *decoder;

/**
* D3D11 VideoContext
*/
ID3D11VideoContext *video_context;

/**
* D3D11 configuration used to create the decoder
*/
D3D11_VIDEO_DECODER_CONFIG *cfg;

/**
* The number of surface in the surface array
*/
unsigned surface_count;

/**
* The array of Direct3D surfaces used to create the decoder
*/
ID3D11VideoDecoderOutputView **surface;

/**
* A bit field configuring the workarounds needed for using the decoder
*/
uint64_t workaround;

/**
* Private to the FFmpeg AVHWAccel implementation
*/
unsigned report_id;

/**
* Mutex to access video_context
*/
HANDLE context_mutex;
} AVD3D11VAContext;

/**
* Allocate an AVD3D11VAContext.
*
* @return Newly-allocated AVD3D11VAContext or NULL on failure.
*/
AVD3D11VAContext *av_d3d11va_alloc_context(void);

/**
* @}
*/

#endif /* AVCODEC_D3D11VA_H */

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ffmpeg/include/libavcodec/defs.h 파일 보기

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/*
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_DEFS_H
#define AVCODEC_DEFS_H

/**
* @file
* @ingroup libavc
* Misc types and constants that do not belong anywhere else.
*/

#include <stdint.h>
#include <stdlib.h>

/**
* @ingroup lavc_decoding
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
* This is mainly needed because some optimized bitstream readers read
* 32 or 64 bit at once and could read over the end.<br>
* Note: If the first 23 bits of the additional bytes are not 0, then damaged
* MPEG bitstreams could cause overread and segfault.
*/
#define AV_INPUT_BUFFER_PADDING_SIZE 64

/**
* Verify checksums embedded in the bitstream (could be of either encoded or
* decoded data, depending on the format) and print an error message on mismatch.
* If AV_EF_EXPLODE is also set, a mismatching checksum will result in the
* decoder/demuxer returning an error.
*/
#define AV_EF_CRCCHECK (1<<0)
#define AV_EF_BITSTREAM (1<<1) ///< detect bitstream specification deviations
#define AV_EF_BUFFER (1<<2) ///< detect improper bitstream length
#define AV_EF_EXPLODE (1<<3) ///< abort decoding on minor error detection

#define AV_EF_IGNORE_ERR (1<<15) ///< ignore errors and continue
#define AV_EF_CAREFUL (1<<16) ///< consider things that violate the spec, are fast to calculate and have not been seen in the wild as errors
#define AV_EF_COMPLIANT (1<<17) ///< consider all spec non compliances as errors
#define AV_EF_AGGRESSIVE (1<<18) ///< consider things that a sane encoder/muxer should not do as an error

#define FF_COMPLIANCE_VERY_STRICT 2 ///< Strictly conform to an older more strict version of the spec or reference software.
#define FF_COMPLIANCE_STRICT 1 ///< Strictly conform to all the things in the spec no matter what consequences.
#define FF_COMPLIANCE_NORMAL 0
#define FF_COMPLIANCE_UNOFFICIAL -1 ///< Allow unofficial extensions
#define FF_COMPLIANCE_EXPERIMENTAL -2 ///< Allow nonstandardized experimental things.

/**
* @ingroup lavc_decoding
*/
enum AVDiscard{
/* We leave some space between them for extensions (drop some
* keyframes for intra-only or drop just some bidir frames). */
AVDISCARD_NONE =-16, ///< discard nothing
AVDISCARD_DEFAULT = 0, ///< discard useless packets like 0 size packets in avi
AVDISCARD_NONREF = 8, ///< discard all non reference
AVDISCARD_BIDIR = 16, ///< discard all bidirectional frames
AVDISCARD_NONINTRA= 24, ///< discard all non intra frames
AVDISCARD_NONKEY = 32, ///< discard all frames except keyframes
AVDISCARD_ALL = 48, ///< discard all
};

enum AVAudioServiceType {
AV_AUDIO_SERVICE_TYPE_MAIN = 0,
AV_AUDIO_SERVICE_TYPE_EFFECTS = 1,
AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED = 2,
AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED = 3,
AV_AUDIO_SERVICE_TYPE_DIALOGUE = 4,
AV_AUDIO_SERVICE_TYPE_COMMENTARY = 5,
AV_AUDIO_SERVICE_TYPE_EMERGENCY = 6,
AV_AUDIO_SERVICE_TYPE_VOICE_OVER = 7,
AV_AUDIO_SERVICE_TYPE_KARAOKE = 8,
AV_AUDIO_SERVICE_TYPE_NB , ///< Not part of ABI
};

/**
* Pan Scan area.
* This specifies the area which should be displayed.
* Note there may be multiple such areas for one frame.
*/
typedef struct AVPanScan {
/**
* id
* - encoding: Set by user.
* - decoding: Set by libavcodec.
*/
int id;

/**
* width and height in 1/16 pel
* - encoding: Set by user.
* - decoding: Set by libavcodec.
*/
int width;
int height;

/**
* position of the top left corner in 1/16 pel for up to 3 fields/frames
* - encoding: Set by user.
* - decoding: Set by libavcodec.
*/
int16_t position[3][2];
} AVPanScan;

/**
* This structure describes the bitrate properties of an encoded bitstream. It
* roughly corresponds to a subset the VBV parameters for MPEG-2 or HRD
* parameters for H.264/HEVC.
*/
typedef struct AVCPBProperties {
/**
* Maximum bitrate of the stream, in bits per second.
* Zero if unknown or unspecified.
*/
int64_t max_bitrate;
/**
* Minimum bitrate of the stream, in bits per second.
* Zero if unknown or unspecified.
*/
int64_t min_bitrate;
/**
* Average bitrate of the stream, in bits per second.
* Zero if unknown or unspecified.
*/
int64_t avg_bitrate;

/**
* The size of the buffer to which the ratecontrol is applied, in bits.
* Zero if unknown or unspecified.
*/
int64_t buffer_size;

/**
* The delay between the time the packet this structure is associated with
* is received and the time when it should be decoded, in periods of a 27MHz
* clock.
*
* UINT64_MAX when unknown or unspecified.
*/
uint64_t vbv_delay;
} AVCPBProperties;

/**
* Allocate a CPB properties structure and initialize its fields to default
* values.
*
* @param size if non-NULL, the size of the allocated struct will be written
* here. This is useful for embedding it in side data.
*
* @return the newly allocated struct or NULL on failure
*/
AVCPBProperties *av_cpb_properties_alloc(size_t *size);

/**
* This structure supplies correlation between a packet timestamp and a wall clock
* production time. The definition follows the Producer Reference Time ('prft')
* as defined in ISO/IEC 14496-12
*/
typedef struct AVProducerReferenceTime {
/**
* A UTC timestamp, in microseconds, since Unix epoch (e.g, av_gettime()).
*/
int64_t wallclock;
int flags;
} AVProducerReferenceTime;

/**
* Encode extradata length to a buffer. Used by xiph codecs.
*
* @param s buffer to write to; must be at least (v/255+1) bytes long
* @param v size of extradata in bytes
* @return number of bytes written to the buffer.
*/
unsigned int av_xiphlacing(unsigned char *s, unsigned int v);

#endif // AVCODEC_DEFS_H

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ffmpeg/include/libavcodec/dirac.h 파일 보기

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/*
* Copyright (C) 2007 Marco Gerards <marco@gnu.org>
* Copyright (C) 2009 David Conrad
* Copyright (C) 2011 Jordi Ortiz
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_DIRAC_H
#define AVCODEC_DIRAC_H

/**
* @file
* Interface to Dirac Decoder/Encoder
* @author Marco Gerards <marco@gnu.org>
* @author David Conrad
* @author Jordi Ortiz
*/

#include "avcodec.h"

/**
* The spec limits the number of wavelet decompositions to 4 for both
* level 1 (VC-2) and 128 (long-gop default).
* 5 decompositions is the maximum before >16-bit buffers are needed.
* Schroedinger allows this for DD 9,7 and 13,7 wavelets only, limiting
* the others to 4 decompositions (or 3 for the fidelity filter).
*
* We use this instead of MAX_DECOMPOSITIONS to save some memory.
*/
#define MAX_DWT_LEVELS 5

/**
* Parse code values:
*
* Dirac Specification ->
* 9.6.1 Table 9.1
*
* VC-2 Specification ->
* 10.4.1 Table 10.1
*/

enum DiracParseCodes {
DIRAC_PCODE_SEQ_HEADER = 0x00,
DIRAC_PCODE_END_SEQ = 0x10,
DIRAC_PCODE_AUX = 0x20,
DIRAC_PCODE_PAD = 0x30,
DIRAC_PCODE_PICTURE_CODED = 0x08,
DIRAC_PCODE_PICTURE_RAW = 0x48,
DIRAC_PCODE_PICTURE_LOW_DEL = 0xC8,
DIRAC_PCODE_PICTURE_HQ = 0xE8,
DIRAC_PCODE_INTER_NOREF_CO1 = 0x0A,
DIRAC_PCODE_INTER_NOREF_CO2 = 0x09,
DIRAC_PCODE_INTER_REF_CO1 = 0x0D,
DIRAC_PCODE_INTER_REF_CO2 = 0x0E,
DIRAC_PCODE_INTRA_REF_CO = 0x0C,
DIRAC_PCODE_INTRA_REF_RAW = 0x4C,
DIRAC_PCODE_INTRA_REF_PICT = 0xCC,
DIRAC_PCODE_MAGIC = 0x42424344,
};

typedef struct DiracVersionInfo {
int major;
int minor;
} DiracVersionInfo;

typedef struct AVDiracSeqHeader {
unsigned width;
unsigned height;
uint8_t chroma_format; ///< 0: 444 1: 422 2: 420

uint8_t interlaced;
uint8_t top_field_first;

uint8_t frame_rate_index; ///< index into dirac_frame_rate[]
uint8_t aspect_ratio_index; ///< index into dirac_aspect_ratio[]

uint16_t clean_width;
uint16_t clean_height;
uint16_t clean_left_offset;
uint16_t clean_right_offset;

uint8_t pixel_range_index; ///< index into dirac_pixel_range_presets[]
uint8_t color_spec_index; ///< index into dirac_color_spec_presets[]

int profile;
int level;

AVRational framerate;
AVRational sample_aspect_ratio;

enum AVPixelFormat pix_fmt;
enum AVColorRange color_range;
enum AVColorPrimaries color_primaries;
enum AVColorTransferCharacteristic color_trc;
enum AVColorSpace colorspace;

DiracVersionInfo version;
int bit_depth;
} AVDiracSeqHeader;

/**
* Parse a Dirac sequence header.
*
* @param dsh this function will allocate and fill an AVDiracSeqHeader struct
* and write it into this pointer. The caller must free it with
* av_free().
* @param buf the data buffer
* @param buf_size the size of the data buffer in bytes
* @param log_ctx if non-NULL, this function will log errors here
* @return 0 on success, a negative AVERROR code on failure
*/
int av_dirac_parse_sequence_header(AVDiracSeqHeader **dsh,
const uint8_t *buf, size_t buf_size,
void *log_ctx);

#endif /* AVCODEC_DIRAC_H */

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ffmpeg/include/libavcodec/dv_profile.h 파일 보기

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_DV_PROFILE_H
#define AVCODEC_DV_PROFILE_H

#include <stdint.h>

#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"

/* minimum number of bytes to read from a DV stream in order to
* determine the profile */
#define DV_PROFILE_BYTES (6 * 80) /* 6 DIF blocks */


/*
* AVDVProfile is used to express the differences between various
* DV flavors. For now it's primarily used for differentiating
* 525/60 and 625/50, but the plans are to use it for various
* DV specs as well (e.g. SMPTE314M vs. IEC 61834).
*/
typedef struct AVDVProfile {
int dsf; /* value of the dsf in the DV header */
int video_stype; /* stype for VAUX source pack */
int frame_size; /* total size of one frame in bytes */
int difseg_size; /* number of DIF segments per DIF channel */
int n_difchan; /* number of DIF channels per frame */
AVRational time_base; /* 1/framerate */
int ltc_divisor; /* FPS from the LTS standpoint */
int height; /* picture height in pixels */
int width; /* picture width in pixels */
AVRational sar[2]; /* sample aspect ratios for 4:3 and 16:9 */
enum AVPixelFormat pix_fmt; /* picture pixel format */
int bpm; /* blocks per macroblock */
const uint8_t *block_sizes; /* AC block sizes, in bits */
int audio_stride; /* size of audio_shuffle table */
int audio_min_samples[3]; /* min amount of audio samples */
/* for 48kHz, 44.1kHz and 32kHz */
int audio_samples_dist[5]; /* how many samples are supposed to be */
/* in each frame in a 5 frames window */
const uint8_t (*audio_shuffle)[9]; /* PCM shuffling table */
} AVDVProfile;

/**
* Get a DV profile for the provided compressed frame.
*
* @param sys the profile used for the previous frame, may be NULL
* @param frame the compressed data buffer
* @param buf_size size of the buffer in bytes
* @return the DV profile for the supplied data or NULL on failure
*/
const AVDVProfile *av_dv_frame_profile(const AVDVProfile *sys,
const uint8_t *frame, unsigned buf_size);

/**
* Get a DV profile for the provided stream parameters.
*/
const AVDVProfile *av_dv_codec_profile(int width, int height, enum AVPixelFormat pix_fmt);

/**
* Get a DV profile for the provided stream parameters.
* The frame rate is used as a best-effort parameter.
*/
const AVDVProfile *av_dv_codec_profile2(int width, int height, enum AVPixelFormat pix_fmt, AVRational frame_rate);

#endif /* AVCODEC_DV_PROFILE_H */

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ffmpeg/include/libavcodec/dxva2.h 파일 보기

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/*
* DXVA2 HW acceleration
*
* copyright (c) 2009 Laurent Aimar
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_DXVA2_H
#define AVCODEC_DXVA2_H

/**
* @file
* @ingroup lavc_codec_hwaccel_dxva2
* Public libavcodec DXVA2 header.
*/

#if !defined(_WIN32_WINNT) || _WIN32_WINNT < 0x0602
#undef _WIN32_WINNT
#define _WIN32_WINNT 0x0602
#endif

#include <stdint.h>
#include <d3d9.h>
#include <dxva2api.h>

/**
* @defgroup lavc_codec_hwaccel_dxva2 DXVA2
* @ingroup lavc_codec_hwaccel
*
* @{
*/

#define FF_DXVA2_WORKAROUND_SCALING_LIST_ZIGZAG 1 ///< Work around for DXVA2 and old UVD/UVD+ ATI video cards
#define FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO 2 ///< Work around for DXVA2 and old Intel GPUs with ClearVideo interface

/**
* This structure is used to provides the necessary configurations and data
* to the DXVA2 FFmpeg HWAccel implementation.
*
* The application must make it available as AVCodecContext.hwaccel_context.
*/
struct dxva_context {
/**
* DXVA2 decoder object
*/
IDirectXVideoDecoder *decoder;

/**
* DXVA2 configuration used to create the decoder
*/
const DXVA2_ConfigPictureDecode *cfg;

/**
* The number of surface in the surface array
*/
unsigned surface_count;

/**
* The array of Direct3D surfaces used to create the decoder
*/
LPDIRECT3DSURFACE9 *surface;

/**
* A bit field configuring the workarounds needed for using the decoder
*/
uint64_t workaround;

/**
* Private to the FFmpeg AVHWAccel implementation
*/
unsigned report_id;
};

/**
* @}
*/

#endif /* AVCODEC_DXVA2_H */

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ffmpeg/include/libavcodec/jni.h 파일 보기

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/*
* JNI public API functions
*
* Copyright (c) 2015-2016 Matthieu Bouron <matthieu.bouron stupeflix.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_JNI_H
#define AVCODEC_JNI_H

/*
* Manually set a Java virtual machine which will be used to retrieve the JNI
* environment. Once a Java VM is set it cannot be changed afterwards, meaning
* you can call multiple times av_jni_set_java_vm with the same Java VM pointer
* however it will error out if you try to set a different Java VM.
*
* @param vm Java virtual machine
* @param log_ctx context used for logging, can be NULL
* @return 0 on success, < 0 otherwise
*/
int av_jni_set_java_vm(void *vm, void *log_ctx);

/*
* Get the Java virtual machine which has been set with av_jni_set_java_vm.
*
* @param vm Java virtual machine
* @return a pointer to the Java virtual machine
*/
void *av_jni_get_java_vm(void *log_ctx);

#endif /* AVCODEC_JNI_H */

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ffmpeg/include/libavcodec/mediacodec.h 파일 보기

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/*
* Android MediaCodec public API
*
* Copyright (c) 2016 Matthieu Bouron <matthieu.bouron stupeflix.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_MEDIACODEC_H
#define AVCODEC_MEDIACODEC_H

#include "libavcodec/avcodec.h"

/**
* This structure holds a reference to a android/view/Surface object that will
* be used as output by the decoder.
*
*/
typedef struct AVMediaCodecContext {

/**
* android/view/Surface object reference.
*/
void *surface;

} AVMediaCodecContext;

/**
* Allocate and initialize a MediaCodec context.
*
* When decoding with MediaCodec is finished, the caller must free the
* MediaCodec context with av_mediacodec_default_free.
*
* @return a pointer to a newly allocated AVMediaCodecContext on success, NULL otherwise
*/
AVMediaCodecContext *av_mediacodec_alloc_context(void);

/**
* Convenience function that sets up the MediaCodec context.
*
* @param avctx codec context
* @param ctx MediaCodec context to initialize
* @param surface reference to an android/view/Surface
* @return 0 on success, < 0 otherwise
*/
int av_mediacodec_default_init(AVCodecContext *avctx, AVMediaCodecContext *ctx, void *surface);

/**
* This function must be called to free the MediaCodec context initialized with
* av_mediacodec_default_init().
*
* @param avctx codec context
*/
void av_mediacodec_default_free(AVCodecContext *avctx);

/**
* Opaque structure representing a MediaCodec buffer to render.
*/
typedef struct MediaCodecBuffer AVMediaCodecBuffer;

/**
* Release a MediaCodec buffer and render it to the surface that is associated
* with the decoder. This function should only be called once on a given
* buffer, once released the underlying buffer returns to the codec, thus
* subsequent calls to this function will have no effect.
*
* @param buffer the buffer to render
* @param render 1 to release and render the buffer to the surface or 0 to
* discard the buffer
* @return 0 on success, < 0 otherwise
*/
int av_mediacodec_release_buffer(AVMediaCodecBuffer *buffer, int render);

/**
* Release a MediaCodec buffer and render it at the given time to the surface
* that is associated with the decoder. The timestamp must be within one second
* of the current `java/lang/System#nanoTime()` (which is implemented using
* `CLOCK_MONOTONIC` on Android). See the Android MediaCodec documentation
* of [`android/media/MediaCodec#releaseOutputBuffer(int,long)`][0] for more details.
*
* @param buffer the buffer to render
* @param time timestamp in nanoseconds of when to render the buffer
* @return 0 on success, < 0 otherwise
*
* [0]: https://developer.android.com/reference/android/media/MediaCodec#releaseOutputBuffer(int,%20long)
*/
int av_mediacodec_render_buffer_at_time(AVMediaCodecBuffer *buffer, int64_t time);

#endif /* AVCODEC_MEDIACODEC_H */

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ffmpeg/include/libavcodec/packet.h 파일 보기

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/*
* AVPacket public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_PACKET_H
#define AVCODEC_PACKET_H

#include <stddef.h>
#include <stdint.h>

#include "libavutil/attributes.h"
#include "libavutil/buffer.h"
#include "libavutil/dict.h"
#include "libavutil/rational.h"
#include "libavutil/version.h"

#include "libavcodec/version_major.h"

/**
* @defgroup lavc_packet AVPacket
*
* Types and functions for working with AVPacket.
* @{
*/
enum AVPacketSideDataType {
/**
* An AV_PKT_DATA_PALETTE side data packet contains exactly AVPALETTE_SIZE
* bytes worth of palette. This side data signals that a new palette is
* present.
*/
AV_PKT_DATA_PALETTE,

/**
* The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format
* that the extradata buffer was changed and the receiving side should
* act upon it appropriately. The new extradata is embedded in the side
* data buffer and should be immediately used for processing the current
* frame or packet.
*/
AV_PKT_DATA_NEW_EXTRADATA,

/**
* An AV_PKT_DATA_PARAM_CHANGE side data packet is laid out as follows:
* @code
* u32le param_flags
* if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_COUNT)
* s32le channel_count
* if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_LAYOUT)
* u64le channel_layout
* if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE)
* s32le sample_rate
* if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_DIMENSIONS)
* s32le width
* s32le height
* @endcode
*/
AV_PKT_DATA_PARAM_CHANGE,

/**
* An AV_PKT_DATA_H263_MB_INFO side data packet contains a number of
* structures with info about macroblocks relevant to splitting the
* packet into smaller packets on macroblock edges (e.g. as for RFC 2190).
* That is, it does not necessarily contain info about all macroblocks,
* as long as the distance between macroblocks in the info is smaller
* than the target payload size.
* Each MB info structure is 12 bytes, and is laid out as follows:
* @code
* u32le bit offset from the start of the packet
* u8 current quantizer at the start of the macroblock
* u8 GOB number
* u16le macroblock address within the GOB
* u8 horizontal MV predictor
* u8 vertical MV predictor
* u8 horizontal MV predictor for block number 3
* u8 vertical MV predictor for block number 3
* @endcode
*/
AV_PKT_DATA_H263_MB_INFO,

/**
* This side data should be associated with an audio stream and contains
* ReplayGain information in form of the AVReplayGain struct.
*/
AV_PKT_DATA_REPLAYGAIN,

/**
* This side data contains a 3x3 transformation matrix describing an affine
* transformation that needs to be applied to the decoded video frames for
* correct presentation.
*
* See libavutil/display.h for a detailed description of the data.
*/
AV_PKT_DATA_DISPLAYMATRIX,

/**
* This side data should be associated with a video stream and contains
* Stereoscopic 3D information in form of the AVStereo3D struct.
*/
AV_PKT_DATA_STEREO3D,

/**
* This side data should be associated with an audio stream and corresponds
* to enum AVAudioServiceType.
*/
AV_PKT_DATA_AUDIO_SERVICE_TYPE,

/**
* This side data contains quality related information from the encoder.
* @code
* u32le quality factor of the compressed frame. Allowed range is between 1 (good) and FF_LAMBDA_MAX (bad).
* u8 picture type
* u8 error count
* u16 reserved
* u64le[error count] sum of squared differences between encoder in and output
* @endcode
*/
AV_PKT_DATA_QUALITY_STATS,

/**
* This side data contains an integer value representing the stream index
* of a "fallback" track. A fallback track indicates an alternate
* track to use when the current track can not be decoded for some reason.
* e.g. no decoder available for codec.
*/
AV_PKT_DATA_FALLBACK_TRACK,

/**
* This side data corresponds to the AVCPBProperties struct.
*/
AV_PKT_DATA_CPB_PROPERTIES,

/**
* Recommmends skipping the specified number of samples
* @code
* u32le number of samples to skip from start of this packet
* u32le number of samples to skip from end of this packet
* u8 reason for start skip
* u8 reason for end skip (0=padding silence, 1=convergence)
* @endcode
*/
AV_PKT_DATA_SKIP_SAMPLES,

/**
* An AV_PKT_DATA_JP_DUALMONO side data packet indicates that
* the packet may contain "dual mono" audio specific to Japanese DTV
* and if it is true, recommends only the selected channel to be used.
* @code
* u8 selected channels (0=main/left, 1=sub/right, 2=both)
* @endcode
*/
AV_PKT_DATA_JP_DUALMONO,

/**
* A list of zero terminated key/value strings. There is no end marker for
* the list, so it is required to rely on the side data size to stop.
*/
AV_PKT_DATA_STRINGS_METADATA,

/**
* Subtitle event position
* @code
* u32le x1
* u32le y1
* u32le x2
* u32le y2
* @endcode
*/
AV_PKT_DATA_SUBTITLE_POSITION,

/**
* Data found in BlockAdditional element of matroska container. There is
* no end marker for the data, so it is required to rely on the side data
* size to recognize the end. 8 byte id (as found in BlockAddId) followed
* by data.
*/
AV_PKT_DATA_MATROSKA_BLOCKADDITIONAL,

/**
* The optional first identifier line of a WebVTT cue.
*/
AV_PKT_DATA_WEBVTT_IDENTIFIER,

/**
* The optional settings (rendering instructions) that immediately
* follow the timestamp specifier of a WebVTT cue.
*/
AV_PKT_DATA_WEBVTT_SETTINGS,

/**
* A list of zero terminated key/value strings. There is no end marker for
* the list, so it is required to rely on the side data size to stop. This
* side data includes updated metadata which appeared in the stream.
*/
AV_PKT_DATA_METADATA_UPDATE,

/**
* MPEGTS stream ID as uint8_t, this is required to pass the stream ID
* information from the demuxer to the corresponding muxer.
*/
AV_PKT_DATA_MPEGTS_STREAM_ID,

/**
* Mastering display metadata (based on SMPTE-2086:2014). This metadata
* should be associated with a video stream and contains data in the form
* of the AVMasteringDisplayMetadata struct.
*/
AV_PKT_DATA_MASTERING_DISPLAY_METADATA,

/**
* This side data should be associated with a video stream and corresponds
* to the AVSphericalMapping structure.
*/
AV_PKT_DATA_SPHERICAL,

/**
* Content light level (based on CTA-861.3). This metadata should be
* associated with a video stream and contains data in the form of the
* AVContentLightMetadata struct.
*/
AV_PKT_DATA_CONTENT_LIGHT_LEVEL,

/**
* ATSC A53 Part 4 Closed Captions. This metadata should be associated with
* a video stream. A53 CC bitstream is stored as uint8_t in AVPacketSideData.data.
* The number of bytes of CC data is AVPacketSideData.size.
*/
AV_PKT_DATA_A53_CC,

/**
* This side data is encryption initialization data.
* The format is not part of ABI, use av_encryption_init_info_* methods to
* access.
*/
AV_PKT_DATA_ENCRYPTION_INIT_INFO,

/**
* This side data contains encryption info for how to decrypt the packet.
* The format is not part of ABI, use av_encryption_info_* methods to access.
*/
AV_PKT_DATA_ENCRYPTION_INFO,

/**
* Active Format Description data consisting of a single byte as specified
* in ETSI TS 101 154 using AVActiveFormatDescription enum.
*/
AV_PKT_DATA_AFD,

/**
* Producer Reference Time data corresponding to the AVProducerReferenceTime struct,
* usually exported by some encoders (on demand through the prft flag set in the
* AVCodecContext export_side_data field).
*/
AV_PKT_DATA_PRFT,

/**
* ICC profile data consisting of an opaque octet buffer following the
* format described by ISO 15076-1.
*/
AV_PKT_DATA_ICC_PROFILE,

/**
* DOVI configuration
* ref:
* dolby-vision-bitstreams-within-the-iso-base-media-file-format-v2.1.2, section 2.2
* dolby-vision-bitstreams-in-mpeg-2-transport-stream-multiplex-v1.2, section 3.3
* Tags are stored in struct AVDOVIDecoderConfigurationRecord.
*/
AV_PKT_DATA_DOVI_CONF,

/**
* Timecode which conforms to SMPTE ST 12-1:2014. The data is an array of 4 uint32_t
* where the first uint32_t describes how many (1-3) of the other timecodes are used.
* The timecode format is described in the documentation of av_timecode_get_smpte_from_framenum()
* function in libavutil/timecode.h.
*/
AV_PKT_DATA_S12M_TIMECODE,

/**
* HDR10+ dynamic metadata associated with a video frame. The metadata is in
* the form of the AVDynamicHDRPlus struct and contains
* information for color volume transform - application 4 of
* SMPTE 2094-40:2016 standard.
*/
AV_PKT_DATA_DYNAMIC_HDR10_PLUS,

/**
* The number of side data types.
* This is not part of the public API/ABI in the sense that it may
* change when new side data types are added.
* This must stay the last enum value.
* If its value becomes huge, some code using it
* needs to be updated as it assumes it to be smaller than other limits.
*/
AV_PKT_DATA_NB
};

#define AV_PKT_DATA_QUALITY_FACTOR AV_PKT_DATA_QUALITY_STATS //DEPRECATED

typedef struct AVPacketSideData {
uint8_t *data;
size_t size;
enum AVPacketSideDataType type;
} AVPacketSideData;

/**
* This structure stores compressed data. It is typically exported by demuxers
* and then passed as input to decoders, or received as output from encoders and
* then passed to muxers.
*
* For video, it should typically contain one compressed frame. For audio it may
* contain several compressed frames. Encoders are allowed to output empty
* packets, with no compressed data, containing only side data
* (e.g. to update some stream parameters at the end of encoding).
*
* The semantics of data ownership depends on the buf field.
* If it is set, the packet data is dynamically allocated and is
* valid indefinitely until a call to av_packet_unref() reduces the
* reference count to 0.
*
* If the buf field is not set av_packet_ref() would make a copy instead
* of increasing the reference count.
*
* The side data is always allocated with av_malloc(), copied by
* av_packet_ref() and freed by av_packet_unref().
*
* sizeof(AVPacket) being a part of the public ABI is deprecated. once
* av_init_packet() is removed, new packets will only be able to be allocated
* with av_packet_alloc(), and new fields may be added to the end of the struct
* with a minor bump.
*
* @see av_packet_alloc
* @see av_packet_ref
* @see av_packet_unref
*/
typedef struct AVPacket {
/**
* A reference to the reference-counted buffer where the packet data is
* stored.
* May be NULL, then the packet data is not reference-counted.
*/
AVBufferRef *buf;
/**
* Presentation timestamp in AVStream->time_base units; the time at which
* the decompressed packet will be presented to the user.
* Can be AV_NOPTS_VALUE if it is not stored in the file.
* pts MUST be larger or equal to dts as presentation cannot happen before
* decompression, unless one wants to view hex dumps. Some formats misuse
* the terms dts and pts/cts to mean something different. Such timestamps
* must be converted to true pts/dts before they are stored in AVPacket.
*/
int64_t pts;
/**
* Decompression timestamp in AVStream->time_base units; the time at which
* the packet is decompressed.
* Can be AV_NOPTS_VALUE if it is not stored in the file.
*/
int64_t dts;
uint8_t *data;
int size;
int stream_index;
/**
* A combination of AV_PKT_FLAG values
*/
int flags;
/**
* Additional packet data that can be provided by the container.
* Packet can contain several types of side information.
*/
AVPacketSideData *side_data;
int side_data_elems;

/**
* Duration of this packet in AVStream->time_base units, 0 if unknown.
* Equals next_pts - this_pts in presentation order.
*/
int64_t duration;

int64_t pos; ///< byte position in stream, -1 if unknown

/**
* for some private data of the user
*/
void *opaque;

/**
* AVBufferRef for free use by the API user. FFmpeg will never check the
* contents of the buffer ref. FFmpeg calls av_buffer_unref() on it when
* the packet is unreferenced. av_packet_copy_props() calls create a new
* reference with av_buffer_ref() for the target packet's opaque_ref field.
*
* This is unrelated to the opaque field, although it serves a similar
* purpose.
*/
AVBufferRef *opaque_ref;

/**
* Time base of the packet's timestamps.
* In the future, this field may be set on packets output by encoders or
* demuxers, but its value will be by default ignored on input to decoders
* or muxers.
*/
AVRational time_base;
} AVPacket;

#if FF_API_INIT_PACKET
attribute_deprecated
typedef struct AVPacketList {
AVPacket pkt;
struct AVPacketList *next;
} AVPacketList;
#endif

#define AV_PKT_FLAG_KEY 0x0001 ///< The packet contains a keyframe
#define AV_PKT_FLAG_CORRUPT 0x0002 ///< The packet content is corrupted
/**
* Flag is used to discard packets which are required to maintain valid
* decoder state but are not required for output and should be dropped
* after decoding.
**/
#define AV_PKT_FLAG_DISCARD 0x0004
/**
* The packet comes from a trusted source.
*
* Otherwise-unsafe constructs such as arbitrary pointers to data
* outside the packet may be followed.
*/
#define AV_PKT_FLAG_TRUSTED 0x0008
/**
* Flag is used to indicate packets that contain frames that can
* be discarded by the decoder. I.e. Non-reference frames.
*/
#define AV_PKT_FLAG_DISPOSABLE 0x0010

enum AVSideDataParamChangeFlags {
#if FF_API_OLD_CHANNEL_LAYOUT
/**
* @deprecated those are not used by any decoder
*/
AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_COUNT = 0x0001,
AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_LAYOUT = 0x0002,
#endif
AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE = 0x0004,
AV_SIDE_DATA_PARAM_CHANGE_DIMENSIONS = 0x0008,
};

/**
* Allocate an AVPacket and set its fields to default values. The resulting
* struct must be freed using av_packet_free().
*
* @return An AVPacket filled with default values or NULL on failure.
*
* @note this only allocates the AVPacket itself, not the data buffers. Those
* must be allocated through other means such as av_new_packet.
*
* @see av_new_packet
*/
AVPacket *av_packet_alloc(void);

/**
* Create a new packet that references the same data as src.
*
* This is a shortcut for av_packet_alloc()+av_packet_ref().
*
* @return newly created AVPacket on success, NULL on error.
*
* @see av_packet_alloc
* @see av_packet_ref
*/
AVPacket *av_packet_clone(const AVPacket *src);

/**
* Free the packet, if the packet is reference counted, it will be
* unreferenced first.
*
* @param pkt packet to be freed. The pointer will be set to NULL.
* @note passing NULL is a no-op.
*/
void av_packet_free(AVPacket **pkt);

#if FF_API_INIT_PACKET
/**
* Initialize optional fields of a packet with default values.
*
* Note, this does not touch the data and size members, which have to be
* initialized separately.
*
* @param pkt packet
*
* @see av_packet_alloc
* @see av_packet_unref
*
* @deprecated This function is deprecated. Once it's removed,
sizeof(AVPacket) will not be a part of the ABI anymore.
*/
attribute_deprecated
void av_init_packet(AVPacket *pkt);
#endif

/**
* Allocate the payload of a packet and initialize its fields with
* default values.
*
* @param pkt packet
* @param size wanted payload size
* @return 0 if OK, AVERROR_xxx otherwise
*/
int av_new_packet(AVPacket *pkt, int size);

/**
* Reduce packet size, correctly zeroing padding
*
* @param pkt packet
* @param size new size
*/
void av_shrink_packet(AVPacket *pkt, int size);

/**
* Increase packet size, correctly zeroing padding
*
* @param pkt packet
* @param grow_by number of bytes by which to increase the size of the packet
*/
int av_grow_packet(AVPacket *pkt, int grow_by);

/**
* Initialize a reference-counted packet from av_malloc()ed data.
*
* @param pkt packet to be initialized. This function will set the data, size,
* and buf fields, all others are left untouched.
* @param data Data allocated by av_malloc() to be used as packet data. If this
* function returns successfully, the data is owned by the underlying AVBuffer.
* The caller may not access the data through other means.
* @param size size of data in bytes, without the padding. I.e. the full buffer
* size is assumed to be size + AV_INPUT_BUFFER_PADDING_SIZE.
*
* @return 0 on success, a negative AVERROR on error
*/
int av_packet_from_data(AVPacket *pkt, uint8_t *data, int size);

/**
* Allocate new information of a packet.
*
* @param pkt packet
* @param type side information type
* @param size side information size
* @return pointer to fresh allocated data or NULL otherwise
*/
uint8_t* av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
size_t size);

/**
* Wrap an existing array as a packet side data.
*
* @param pkt packet
* @param type side information type
* @param data the side data array. It must be allocated with the av_malloc()
* family of functions. The ownership of the data is transferred to
* pkt.
* @param size side information size
* @return a non-negative number on success, a negative AVERROR code on
* failure. On failure, the packet is unchanged and the data remains
* owned by the caller.
*/
int av_packet_add_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
uint8_t *data, size_t size);

/**
* Shrink the already allocated side data buffer
*
* @param pkt packet
* @param type side information type
* @param size new side information size
* @return 0 on success, < 0 on failure
*/
int av_packet_shrink_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
size_t size);

/**
* Get side information from packet.
*
* @param pkt packet
* @param type desired side information type
* @param size If supplied, *size will be set to the size of the side data
* or to zero if the desired side data is not present.
* @return pointer to data if present or NULL otherwise
*/
uint8_t* av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type,
size_t *size);

const char *av_packet_side_data_name(enum AVPacketSideDataType type);

/**
* Pack a dictionary for use in side_data.
*
* @param dict The dictionary to pack.
* @param size pointer to store the size of the returned data
* @return pointer to data if successful, NULL otherwise
*/
uint8_t *av_packet_pack_dictionary(AVDictionary *dict, size_t *size);
/**
* Unpack a dictionary from side_data.
*
* @param data data from side_data
* @param size size of the data
* @param dict the metadata storage dictionary
* @return 0 on success, < 0 on failure
*/
int av_packet_unpack_dictionary(const uint8_t *data, size_t size,
AVDictionary **dict);

/**
* Convenience function to free all the side data stored.
* All the other fields stay untouched.
*
* @param pkt packet
*/
void av_packet_free_side_data(AVPacket *pkt);

/**
* Setup a new reference to the data described by a given packet
*
* If src is reference-counted, setup dst as a new reference to the
* buffer in src. Otherwise allocate a new buffer in dst and copy the
* data from src into it.
*
* All the other fields are copied from src.
*
* @see av_packet_unref
*
* @param dst Destination packet. Will be completely overwritten.
* @param src Source packet
*
* @return 0 on success, a negative AVERROR on error. On error, dst
* will be blank (as if returned by av_packet_alloc()).
*/
int av_packet_ref(AVPacket *dst, const AVPacket *src);

/**
* Wipe the packet.
*
* Unreference the buffer referenced by the packet and reset the
* remaining packet fields to their default values.
*
* @param pkt The packet to be unreferenced.
*/
void av_packet_unref(AVPacket *pkt);

/**
* Move every field in src to dst and reset src.
*
* @see av_packet_unref
*
* @param src Source packet, will be reset
* @param dst Destination packet
*/
void av_packet_move_ref(AVPacket *dst, AVPacket *src);

/**
* Copy only "properties" fields from src to dst.
*
* Properties for the purpose of this function are all the fields
* beside those related to the packet data (buf, data, size)
*
* @param dst Destination packet
* @param src Source packet
*
* @return 0 on success AVERROR on failure.
*/
int av_packet_copy_props(AVPacket *dst, const AVPacket *src);

/**
* Ensure the data described by a given packet is reference counted.
*
* @note This function does not ensure that the reference will be writable.
* Use av_packet_make_writable instead for that purpose.
*
* @see av_packet_ref
* @see av_packet_make_writable
*
* @param pkt packet whose data should be made reference counted.
*
* @return 0 on success, a negative AVERROR on error. On failure, the
* packet is unchanged.
*/
int av_packet_make_refcounted(AVPacket *pkt);

/**
* Create a writable reference for the data described by a given packet,
* avoiding data copy if possible.
*
* @param pkt Packet whose data should be made writable.
*
* @return 0 on success, a negative AVERROR on failure. On failure, the
* packet is unchanged.
*/
int av_packet_make_writable(AVPacket *pkt);

/**
* Convert valid timing fields (timestamps / durations) in a packet from one
* timebase to another. Timestamps with unknown values (AV_NOPTS_VALUE) will be
* ignored.
*
* @param pkt packet on which the conversion will be performed
* @param tb_src source timebase, in which the timing fields in pkt are
* expressed
* @param tb_dst destination timebase, to which the timing fields will be
* converted
*/
void av_packet_rescale_ts(AVPacket *pkt, AVRational tb_src, AVRational tb_dst);

/**
* @}
*/

#endif // AVCODEC_PACKET_H

+ 109
- 0
ffmpeg/include/libavcodec/qsv.h 파일 보기

@@ -0,0 +1,109 @@
/*
* Intel MediaSDK QSV public API
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_QSV_H
#define AVCODEC_QSV_H

#include <mfxvideo.h>

#include "libavutil/buffer.h"

/**
* This struct is used for communicating QSV parameters between libavcodec and
* the caller. It is managed by the caller and must be assigned to
* AVCodecContext.hwaccel_context.
* - decoding: hwaccel_context must be set on return from the get_format()
* callback
* - encoding: hwaccel_context must be set before avcodec_open2()
*/
typedef struct AVQSVContext {
/**
* If non-NULL, the session to use for encoding or decoding.
* Otherwise, libavcodec will try to create an internal session.
*/
mfxSession session;

/**
* The IO pattern to use.
*/
int iopattern;

/**
* Extra buffers to pass to encoder or decoder initialization.
*/
mfxExtBuffer **ext_buffers;
int nb_ext_buffers;

/**
* Encoding only. If this field is set to non-zero by the caller, libavcodec
* will create an mfxExtOpaqueSurfaceAlloc extended buffer and pass it to
* the encoder initialization. This only makes sense if iopattern is also
* set to MFX_IOPATTERN_IN_OPAQUE_MEMORY.
*
* The number of allocated opaque surfaces will be the sum of the number
* required by the encoder and the user-provided value nb_opaque_surfaces.
* The array of the opaque surfaces will be exported to the caller through
* the opaque_surfaces field.
*
* The caller must set this field to zero for oneVPL (MFX_VERSION >= 2.0)
*/
int opaque_alloc;

/**
* Encoding only, and only if opaque_alloc is set to non-zero. Before
* calling avcodec_open2(), the caller should set this field to the number
* of extra opaque surfaces to allocate beyond what is required by the
* encoder.
*
* On return from avcodec_open2(), this field will be set by libavcodec to
* the total number of allocated opaque surfaces.
*/
int nb_opaque_surfaces;

/**
* Encoding only, and only if opaque_alloc is set to non-zero. On return
* from avcodec_open2(), this field will be used by libavcodec to export the
* array of the allocated opaque surfaces to the caller, so they can be
* passed to other parts of the pipeline.
*
* The buffer reference exported here is owned and managed by libavcodec,
* the callers should make their own reference with av_buffer_ref() and free
* it with av_buffer_unref() when it is no longer needed.
*
* The buffer data is an nb_opaque_surfaces-sized array of mfxFrameSurface1.
*/
AVBufferRef *opaque_surfaces;

/**
* Encoding only, and only if opaque_alloc is set to non-zero. On return
* from avcodec_open2(), this field will be set to the surface type used in
* the opaque allocation request.
*/
int opaque_alloc_type;
} AVQSVContext;

/**
* Allocate a new context.
*
* It must be freed by the caller with av_free().
*/
AVQSVContext *av_qsv_alloc_context(void);

#endif /* AVCODEC_QSV_H */

+ 157
- 0
ffmpeg/include/libavcodec/vdpau.h 파일 보기

@@ -0,0 +1,157 @@
/*
* The Video Decode and Presentation API for UNIX (VDPAU) is used for
* hardware-accelerated decoding of MPEG-1/2, H.264 and VC-1.
*
* Copyright (C) 2008 NVIDIA
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_VDPAU_H
#define AVCODEC_VDPAU_H

/**
* @file
* @ingroup lavc_codec_hwaccel_vdpau
* Public libavcodec VDPAU header.
*/


/**
* @defgroup lavc_codec_hwaccel_vdpau VDPAU Decoder and Renderer
* @ingroup lavc_codec_hwaccel
*
* VDPAU hardware acceleration has two modules
* - VDPAU decoding
* - VDPAU presentation
*
* The VDPAU decoding module parses all headers using FFmpeg
* parsing mechanisms and uses VDPAU for the actual decoding.
*
* As per the current implementation, the actual decoding
* and rendering (API calls) are done as part of the VDPAU
* presentation (vo_vdpau.c) module.
*
* @{
*/

#include <vdpau/vdpau.h>

#include "libavutil/avconfig.h"
#include "libavutil/attributes.h"

#include "avcodec.h"

struct AVCodecContext;
struct AVFrame;

typedef int (*AVVDPAU_Render2)(struct AVCodecContext *, struct AVFrame *,
const VdpPictureInfo *, uint32_t,
const VdpBitstreamBuffer *);

/**
* This structure is used to share data between the libavcodec library and
* the client video application.
* The user shall allocate the structure via the av_alloc_vdpau_hwaccel
* function and make it available as
* AVCodecContext.hwaccel_context. Members can be set by the user once
* during initialization or through each AVCodecContext.get_buffer()
* function call. In any case, they must be valid prior to calling
* decoding functions.
*
* The size of this structure is not a part of the public ABI and must not
* be used outside of libavcodec. Use av_vdpau_alloc_context() to allocate an
* AVVDPAUContext.
*/
typedef struct AVVDPAUContext {
/**
* VDPAU decoder handle
*
* Set by user.
*/
VdpDecoder decoder;

/**
* VDPAU decoder render callback
*
* Set by the user.
*/
VdpDecoderRender *render;

AVVDPAU_Render2 render2;
} AVVDPAUContext;

/**
* @brief allocation function for AVVDPAUContext
*
* Allows extending the struct without breaking API/ABI
*/
AVVDPAUContext *av_alloc_vdpaucontext(void);

AVVDPAU_Render2 av_vdpau_hwaccel_get_render2(const AVVDPAUContext *);
void av_vdpau_hwaccel_set_render2(AVVDPAUContext *, AVVDPAU_Render2);

/**
* Associate a VDPAU device with a codec context for hardware acceleration.
* This function is meant to be called from the get_format() codec callback,
* or earlier. It can also be called after avcodec_flush_buffers() to change
* the underlying VDPAU device mid-stream (e.g. to recover from non-transparent
* display preemption).
*
* @note get_format() must return AV_PIX_FMT_VDPAU if this function completes
* successfully.
*
* @param avctx decoding context whose get_format() callback is invoked
* @param device VDPAU device handle to use for hardware acceleration
* @param get_proc_address VDPAU device driver
* @param flags zero of more OR'd AV_HWACCEL_FLAG_* flags
*
* @return 0 on success, an AVERROR code on failure.
*/
int av_vdpau_bind_context(AVCodecContext *avctx, VdpDevice device,
VdpGetProcAddress *get_proc_address, unsigned flags);

/**
* Gets the parameters to create an adequate VDPAU video surface for the codec
* context using VDPAU hardware decoding acceleration.
*
* @note Behavior is undefined if the context was not successfully bound to a
* VDPAU device using av_vdpau_bind_context().
*
* @param avctx the codec context being used for decoding the stream
* @param type storage space for the VDPAU video surface chroma type
* (or NULL to ignore)
* @param width storage space for the VDPAU video surface pixel width
* (or NULL to ignore)
* @param height storage space for the VDPAU video surface pixel height
* (or NULL to ignore)
*
* @return 0 on success, a negative AVERROR code on failure.
*/
int av_vdpau_get_surface_parameters(AVCodecContext *avctx, VdpChromaType *type,
uint32_t *width, uint32_t *height);

/**
* Allocate an AVVDPAUContext.
*
* @return Newly-allocated AVVDPAUContext or NULL on failure.
*/
AVVDPAUContext *av_vdpau_alloc_context(void);

/** @} */

#endif /* AVCODEC_VDPAU_H */

+ 45
- 0
ffmpeg/include/libavcodec/version.h 파일 보기

@@ -0,0 +1,45 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_VERSION_H
#define AVCODEC_VERSION_H

/**
* @file
* @ingroup libavc
* Libavcodec version macros.
*/

#include "libavutil/version.h"

#include "version_major.h"

#define LIBAVCODEC_VERSION_MINOR 6
#define LIBAVCODEC_VERSION_MICRO 101

#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
LIBAVCODEC_VERSION_MICRO)
#define LIBAVCODEC_VERSION AV_VERSION(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
LIBAVCODEC_VERSION_MICRO)
#define LIBAVCODEC_BUILD LIBAVCODEC_VERSION_INT

#define LIBAVCODEC_IDENT "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)

#endif /* AVCODEC_VERSION_H */

+ 53
- 0
ffmpeg/include/libavcodec/version_major.h 파일 보기

@@ -0,0 +1,53 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_VERSION_MAJOR_H
#define AVCODEC_VERSION_MAJOR_H

/**
* @file
* @ingroup libavc
* Libavcodec version macros.
*/

#define LIBAVCODEC_VERSION_MAJOR 60

/**
* FF_API_* defines may be placed below to indicate public API that will be
* dropped at a future version bump. The defines themselves are not part of
* the public API and may change, break or disappear at any time.
*
* @note, when bumping the major version it is recommended to manually
* disable each FF_API_* in its own commit instead of disabling them all
* at once through the bump. This improves the git bisect-ability of the change.
*/

#define FF_API_INIT_PACKET (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_IDCT_NONE (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_SVTAV1_OPTS (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_AYUV_CODECID (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_VT_OUTPUT_CALLBACK (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_AVCODEC_CHROMA_POS (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_VT_HWACCEL_CONTEXT (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_AVCTX_FRAME_NUMBER (LIBAVCODEC_VERSION_MAJOR < 61)
#define FF_API_SLICE_OFFSET (LIBAVCODEC_VERSION_MAJOR < 61)

// reminder to remove CrystalHD decoders on next major bump
#define FF_CODEC_CRYSTAL_HD (LIBAVCODEC_VERSION_MAJOR < 61)

#endif /* AVCODEC_VERSION_MAJOR_H */

+ 150
- 0
ffmpeg/include/libavcodec/videotoolbox.h 파일 보기

@@ -0,0 +1,150 @@
/*
* Videotoolbox hardware acceleration
*
* copyright (c) 2012 Sebastien Zwickert
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#ifndef AVCODEC_VIDEOTOOLBOX_H
#define AVCODEC_VIDEOTOOLBOX_H

/**
* @file
* @ingroup lavc_codec_hwaccel_videotoolbox
* Public libavcodec Videotoolbox header.
*/

/**
* @defgroup lavc_codec_hwaccel_videotoolbox VideoToolbox Decoder
* @ingroup lavc_codec_hwaccel
*
* Hardware accelerated decoding using VideoToolbox on Apple Platforms
*
* @{
*/

#include <stdint.h>

#define Picture QuickdrawPicture
#include <VideoToolbox/VideoToolbox.h>
#undef Picture

#include "libavcodec/avcodec.h"

#include "libavutil/attributes.h"

/**
* This struct holds all the information that needs to be passed
* between the caller and libavcodec for initializing Videotoolbox decoding.
* Its size is not a part of the public ABI, it must be allocated with
* av_videotoolbox_alloc_context() and freed with av_free().
*/
typedef struct AVVideotoolboxContext {
/**
* Videotoolbox decompression session object.
*/
VTDecompressionSessionRef session;

#if FF_API_VT_OUTPUT_CALLBACK
/**
* The output callback that must be passed to the session.
* Set by av_videottoolbox_default_init()
*/
attribute_deprecated
VTDecompressionOutputCallback output_callback;
#endif

/**
* CVPixelBuffer Format Type that Videotoolbox will use for decoded frames.
* set by the caller. If this is set to 0, then no specific format is
* requested from the decoder, and its native format is output.
*/
OSType cv_pix_fmt_type;

/**
* CoreMedia Format Description that Videotoolbox will use to create the decompression session.
*/
CMVideoFormatDescriptionRef cm_fmt_desc;

/**
* CoreMedia codec type that Videotoolbox will use to create the decompression session.
*/
int cm_codec_type;
} AVVideotoolboxContext;

#if FF_API_VT_HWACCEL_CONTEXT

/**
* Allocate and initialize a Videotoolbox context.
*
* This function should be called from the get_format() callback when the caller
* selects the AV_PIX_FMT_VIDETOOLBOX format. The caller must then create
* the decoder object (using the output callback provided by libavcodec) that
* will be used for Videotoolbox-accelerated decoding.
*
* When decoding with Videotoolbox is finished, the caller must destroy the decoder
* object and free the Videotoolbox context using av_free().
*
* @return the newly allocated context or NULL on failure
* @deprecated Use AVCodecContext.hw_frames_ctx or hw_device_ctx instead.
*/
attribute_deprecated
AVVideotoolboxContext *av_videotoolbox_alloc_context(void);

/**
* This is a convenience function that creates and sets up the Videotoolbox context using
* an internal implementation.
*
* @param avctx the corresponding codec context
*
* @return >= 0 on success, a negative AVERROR code on failure
* @deprecated Use AVCodecContext.hw_frames_ctx or hw_device_ctx instead.
*/
attribute_deprecated
int av_videotoolbox_default_init(AVCodecContext *avctx);

/**
* This is a convenience function that creates and sets up the Videotoolbox context using
* an internal implementation.
*
* @param avctx the corresponding codec context
* @param vtctx the Videotoolbox context to use
*
* @return >= 0 on success, a negative AVERROR code on failure
* @deprecated Use AVCodecContext.hw_frames_ctx or hw_device_ctx instead.
*/
attribute_deprecated
int av_videotoolbox_default_init2(AVCodecContext *avctx, AVVideotoolboxContext *vtctx);

/**
* This function must be called to free the Videotoolbox context initialized with
* av_videotoolbox_default_init().
*
* @param avctx the corresponding codec context
* @deprecated Use AVCodecContext.hw_frames_ctx or hw_device_ctx instead.
*/
attribute_deprecated
void av_videotoolbox_default_free(AVCodecContext *avctx);

#endif /* FF_API_VT_HWACCEL_CONTEXT */

/**
* @}
*/

#endif /* AVCODEC_VIDEOTOOLBOX_H */

+ 74
- 0
ffmpeg/include/libavcodec/vorbis_parser.h 파일 보기

@@ -0,0 +1,74 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

/**
* @file
* A public API for Vorbis parsing
*
* Determines the duration for each packet.
*/

#ifndef AVCODEC_VORBIS_PARSER_H
#define AVCODEC_VORBIS_PARSER_H

#include <stdint.h>

typedef struct AVVorbisParseContext AVVorbisParseContext;

/**
* Allocate and initialize the Vorbis parser using headers in the extradata.
*/
AVVorbisParseContext *av_vorbis_parse_init(const uint8_t *extradata,
int extradata_size);

/**
* Free the parser and everything associated with it.
*/
void av_vorbis_parse_free(AVVorbisParseContext **s);

#define VORBIS_FLAG_HEADER 0x00000001
#define VORBIS_FLAG_COMMENT 0x00000002
#define VORBIS_FLAG_SETUP 0x00000004

/**
* Get the duration for a Vorbis packet.
*
* If @p flags is @c NULL,
* special frames are considered invalid.
*
* @param s Vorbis parser context
* @param buf buffer containing a Vorbis frame
* @param buf_size size of the buffer
* @param flags flags for special frames
*/
int av_vorbis_parse_frame_flags(AVVorbisParseContext *s, const uint8_t *buf,
int buf_size, int *flags);

/**
* Get the duration for a Vorbis packet.
*
* @param s Vorbis parser context
* @param buf buffer containing a Vorbis frame
* @param buf_size size of the buffer
*/
int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf,
int buf_size);

void av_vorbis_parse_reset(AVVorbisParseContext *s);

#endif /* AVCODEC_VORBIS_PARSER_H */

+ 0
- 0
ffmpeg/include/libavcodec/xvmc.h 파일 보기


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