ffmpeg与ffserver的协同工作

╰+攻爆jí腚メ 2022-05-16 21:17 150阅读 0赞

ffmpeg和ffserver配合使用可以实现实时的流媒体服务，可以实时传输来自摄像头的数据，客户端可以采用HTTP、RTSP、RTP协议等播放视频流。

**一、概念和流程**

ffmpeg和ffserver配合使用涉及到四个概念：

1. **ffmpeg**，负责媒体文件的转码工作，把你服务器上的源媒体文件转换成要发送出去的流媒体文件。

2. **ffserver**，负责响应客户端的流媒体请求，把流媒体数据发送给客户端。

3. **ffserver.conf**，ffserver启动时的配置文件，在这个文件中主要是对网络协议，缓存文件feed1.ffm和要发送的流媒体文件的格式参数做具体的设定。

4. **feed1.ffm**，可以看成是一个流媒体数据的缓存文件，ffmpeg把转码好的数据发送给ffserver，如果没有客户端连接请求，ffserver把数据缓存到该文件中。

**工作流程**如下：

**1、 启动ffserver，配置参数**

ffserver先于ffmpeg启动，它在启动的时候需要加参数-f指定其配置文件，配置文件里包含端口信息、缓冲文件配置、传送流配置(如编码方式，帧率，采样率……)。

**2、 启动ffmpeg，输入流**

启动ffmpeg，向缓冲文件输入数据流，数据流可以来自摄像头，也可以来自本来就存在的文件。

feed1.ffm是一个缓冲文件，fserver启动后，feed1.ffm就会自动被创建，feed1.ffm开始的部分已经写入向客户端传送流的配置信息，在feed1.ffm做缓冲用的时候，这些信息是不会被覆盖掉。

ffmpeg启动的一个关键参数就是“http://ip:port/feed1.ffm”，其中ip是运行ffserver主机的ip，如果ffmpeg和ffserver都在同一系统中运行的话，用localhost或者127.0.0.1也行。ffmpeg启动后会与ffserver建立一个连接(短暂的连接)，通过这第一次的连接，ffmpeg**从ffserver那里获取了向客户端输出流的配置，并把这些配置作为自己编码输出的配置**，然后ffmpeg断开了这次连接，再次与ffserver建立连接(长久的连接)，利用这个连接ffmpeg会把编码后的数据发送给ffserver。如果你观察ffserver端的输出就会发现这段时间会出现两次HTTP的200，这就是两次连接的过程。

**3、连接过程**

ffmpeg从摄像头获取数据后，按照输出流的编码方式编码，然后发送给ffserver，ffserver收到ffmpeg的数据后，如果网络上没有播放的请求，就把数据写入feed1.ffm中缓存，写入时把数据加上些头信息然后分块，每块4096B（每块也有结构），当feed1.ffm的大小到了ffserver.conf中规定的大小后，就会从文件开始（跳过头）写入，覆盖旧的数据。直到网络上有播放的请求，**ffserver从feed1.ffm中读取数据，发送给客户端**。

**二、配置与应用**

ffserver可以配置为带缓冲或者不带缓冲，其中不带缓冲的只需要配置stream的位置，不需要feed和ffmpeg。

ffserver配置文件可以参考ffmpeg源码中的doc/ffserver.conf，里边有详细的注释。文件的结构可以分为头部信息、实时流信息、格式信息。

**1、不带缓冲**

最简单的配置文件如下

Port 9999

RTSPPort9990

BindAddress0.0.0.0

MaxClients1000

MaxBandwidth100000

CustomLog–

\#只需要指定待播放的文件的路径以及格式信息即可

<Streamtest.flv>

File "/home/test.flv"

Format flv

</Stream>

\#rtsp应用

<Streamtest.mpg>

File"myfile/testvideo/test.mpg"

Format rtp

</Stream>

**命令符：**

1. 在终端里输入ffserver -f /etc/ffserver.conf

2. 在浏览器里或者相关播放器地址里输入 http://ipAddr:port/test.flv

**备注**：1、Port为配置里面的9999，文件名直接输入流的文件名即可。

2、实际测试flv等格式都可以播放。

3、测试需要的test.flv的可以使用ffmpeg录制，命令是

ffmpeg -f v4l2 -s 320\*240 -r 10 -i /dev/video2-vcodec flv /test.flv

**2、带缓冲**

配置文件如下

Port 9999

RTSPPort9990

BindAddress0.0.0.0

MaxClients1000

MaxBandwidth10000

CustomLog–

File/tmp/feed1.ffm

FileMaxSize40k

ACL allow127.0.0.1

</Feed>

Feedfeed1.ffm

Formatflv

BitExact

DctFastint

IdctSimple

VideoFrameRate10

VideoSize320x240

VideoBitRate64

VideoGopSize10

NoAudio

PreRoll10

StartSendOnKey

MaxTime100

</Stream>

**要点：**

1、实时流数据配置，其中注意文件的位置，可以放到tmp文件夹下面，这样会被自动清理掉。

2、每个不同的流都来自feed1.ffm，因此配置越多的流，当执行的时候，会逐个转换，影响速度，一般不建议多配置。

3、ACL allow表示ip的地址范围，比如ACL allow 192.168.0.0 192.168.255.255

**命令符：**

1． 在终端里输入

ffserver -f/etc/ffserver.conf

2． a.若是文件方式则输入

ffmpeg -i/home/test.flv http://127.0.0.1:9999/test.flv

b.若是实时视频则输入

ffmpeg -fv4l2 -framerate 30 -i /dev/video2[http://127.0.0.1:9999/feed1.ffm][http_127.0.0.1_9999_feed1.ffm]

3、运行客户端命令

[http://192.168.1.230:9999/test.flv][http_192.168.1.230_9999_test.flv]

rtsp://ip:port/rtsp.mpg

**三、流的格式**

文件的拓展名对应一定的格式，常用的有：

配置例子：

Multipart JPEG

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.mjpg&gt; Feed feed1.ffm Format mpjpeg VideoFrameRate 2 VideoIntraOnly NoAudio Strict -1 &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

Single JPEG

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.jpg&gt; Feed feed1.ffm Format jpeg VideoFrameRate 2 VideoIntraOnly VideoSize 352x240 NoAudio Strict -1 &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

Flash

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.swf&gt; Feed feed1.ffm Format swf VideoFrameRate 2 VideoIntraOnly NoAudio &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

ASF compatible

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.asf&gt; Feed feed1.ffm Format asf VideoFrameRate 15 VideoSize 352x240 VideoBitRate 256 VideoBufferSize 40 VideoGopSize 30 AudioBitRate 64 StartSendOnKey &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

MP3 audio

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.mp3&gt; Feed feed1.ffm Format mp2 AudioCodec mp3 AudioBitRate 64 AudioChannels 1 AudioSampleRate 44100 NoVideo &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

Ogg Vorbis audio

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.ogg&gt; Feed feed1.ffm Metadata title "Stream title" AudioBitRate 64 AudioChannels 2 AudioSampleRate 44100 NoVideo &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

Real with audio only at 32 kbits

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.ra&gt; Feed feed1.ffm Format rm AudioBitRate 32 NoVideo &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

Real with audio and video at 64 kbits

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream test.rm&gt; Feed feed1.ffm Format rm AudioBitRate 32 VideoBitRate 128 VideoFrameRate 25 VideoGopSize 25 &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

For stream coming from a file: you onlyneed to set the input filename and optionally a new format.

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream file.rm&gt; File "/usr/local/httpd/htdocs/tlive.rm" NoAudio &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

<table> 
 <tbody> 
 <tr> 
 <td> &nbsp; </td> 
 <td> &lt;Stream file.asf&gt; File "/usr/local/httpd/htdocs/test.asf" NoAudio Metadata author "Me" Metadata copyright "Super MegaCorp" Metadata title "Test stream from disk" Metadata comment "Test comment" &lt;/Stream&gt; </td> 
 </tr> 
 </tbody> 
</table>

实测可行的例子

配置：

<Streammy.mp4>

Formatrtp

File"/home/my.mp4"

</Stream>

客户端命令 rtsp://192.168.1.230:9990/my.mp4

端口就是rtp的端口。使用http协议不能访问。

<Streamtest.mp4>

Feedfeed1.ffm

Formatrtp

BitExact

DctFastint

IdctSimple

VideoFrameRate10

VideoSize320x240

VideoBitRate64

VideoGopSize10

NoAudio

PreRoll10

StartSendOnKey

MaxTime100

</Stream>

ffmpeg -fv4l2 -r 10 -I /dev/video2 [http://127.0.0.1:9999/feed1.ffm][http_127.0.0.1_9999_feed1.ffm]

客户端命令 rtsp://192.168.1.230:9990/test.mp4

<Streamlive.h264>

Formatrtp

Feedfeed1.ffm

VideoCodeclibx264

VideoFrameRate24

VideoBitRate100

VideoSize480x272

AVPresetVideodefault

AVPresetVideobaseline

AVOptionVideoflags +global\_header

AudioCodeclibfaac

AudioBitRate32

AudioChannels2

AudioSampleRate22050

AVOptionAudioflags +global\_header

</Stream>

使用H.264编码时，使用命令

ffmpeg -f v4l2 -s 176\*144 -r 2 -i /dev/video0-vcodec libx264 http://192.168.1.6:8090/feed1.ffm

ffmpeg-f v4l2 -s 176\*144 -r 2 **-vpre libx264-hq.ffpreset** \-i /dev/video0 -vcodeclibx264[http://localhost:8090/feed1.ffm][http_localhost_8090_feed1.ffm]

ffmpeg-f v4l2 -s 176\*144 -r 10 -vpre libx264-hq.ffpreset-i /dev/video0 -vcodec libx264 -f rtprtp://192.168.1.105:6060 > /tmp/x264.sdp

四、协议

HTTP协议的地址格式为：

http://*ffserver\_ip\_address*:*http\_port*/*stream\_name*\[*options*\]

RTSP协议的地址格式为：

http://*ffserver\_ip\_address*:*rtsp\_port*/*stream\_name*\[*options*\]

# Sampleffserver configuration file #

Port 8090

BindAddress 0.0.0.0

MaxHTTPConnections 2000

MaxClients 1000

MaxBandwidth 1000

CustomLog -

##################################################################

# ffmpeg http://localhost:8090/feed1.ffm

File /tmp/feed1.ffm

FileMaxSize 200K

ACL allow 127.0.0.1

</Feed>

##################################################################

Feed feed1.ffm

# Format of the stream : you can choose among:

# mpeg       : MPEG-1 multiplexed video and audio

# mpegvideo  : only MPEG-1 video

# mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)

# ogg        : Ogg format (Vorbis audio codec)

# rm         : RealNetworks-compatible stream. Multiplexed audio and video.

# ra         : RealNetworks-compatible stream. Audio only.

# mpjpeg     : Multipart JPEG (works with Netscape without any plugin)

# jpeg       : Generate a single JPEG image.

# asf        : ASF compatible streaming (Windows Media Player format).

# swf        : Macromedia Flash compatible stream

# avi        : AVI format (MPEG-4 video, MPEG audio sound)

Format mpeg

# Bitrate for the audio stream. Codecs usually support only a few

# different bitrates.

AudioBitRate 32

# Number of audio channels: 1 = mono, 2 = stereo

AudioChannels 1

# Sampling frequency for audio. When using low bitrates, you should

# lower this frequency to 22050 or 11025. The supported frequencies

# depend on the selected audio codec.

AudioSampleRate 44100

# Bitrate for the video stream

VideoBitRate 64

# Ratecontrol buffer size

VideoBufferSize 40

# Number of frames per second

VideoFrameRate 3

# Size of the video frame: WxH (default: 160x128)

# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,

# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,

# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,

# hd1080

VideoSize 160x128

# Transmit only intra frames (useful for low bitrates, but kills frame rate).

#VideoIntraOnly

# If non-intra only, an intra frame is transmitted every VideoGopSize

# frames. Video synchronization can only begin at an intra frame.

VideoGopSize 12

# More MPEG-4 parameters

# VideoHighQuality

# Video4MotionVector

# Choose your codecs:

#AudioCodec mp2

#VideoCodec mpeg1video

# Suppress audio

#NoAudio

# Suppress video

#NoVideo

#VideoQMin 3

#VideoQMax 31

# Set this to the number of seconds backwards in time to start. Note that

# most players will buffer 5-10 seconds of video, and also you need to allow

# for a keyframe to appear in the data stream.

#Preroll 15

# ACL:

# You can allow ranges of addresses (or single addresses)

#ACL ALLOW <first address>

# You can deny ranges of addresses (or single addresses)

#ACL DENY <first address>

# You can repeat the ACL allow/deny as often as you like. It is on a per

# stream basis. The first match defines the action. If there are no matches,

# then the default is the inverse of the last ACL statement.

# Thus 'ACL allow localhost' only allows access from localhost.

# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and

# allow everybody else.

</Stream>

##################################################################

# Example streams

# Multipart JPEG

#<Stream test.mjpg>

#Feed feed1.ffm

#Format mpjpeg

#VideoFrameRate 2

#VideoIntraOnly

#NoAudio

#Strict -1

#</Stream>

# Single JPEG

#<Stream test.jpg>

#Feed feed1.ffm

#Format jpeg

#VideoFrameRate 2

#VideoIntraOnly

##VideoSize 352x240

#NoAudio

#Strict -1

#</Stream>

# Flash

#<Stream test.swf>

#Feed feed1.ffm

#Format swf

#VideoFrameRate 2

#VideoIntraOnly

#NoAudio

#</Stream>

# ASF compatible

Feed feed1.ffm

Format asf

VideoFrameRate 15

VideoSize 352x240

VideoBitRate 256

VideoBufferSize 40

VideoGopSize 30

AudioBitRate 64

StartSendOnKey

</Stream>

# MP3 audio

#<Stream test.mp3>

#Feed feed1.ffm

#Format mp2

#AudioCodec mp3

#AudioBitRate 64

#AudioChannels 1

#AudioSampleRate 44100

#NoVideo

#</Stream>

# Ogg Vorbis audio

#<Stream test.ogg>

#Feed feed1.ffm

#Title "Stream title"

#AudioBitRate 64

#AudioChannels 2

#AudioSampleRate 44100

#NoVideo

#</Stream>

# Real with audio only at 32 kbits

#<Stream test.ra>

#Feed feed1.ffm

#Format rm

#AudioBitRate 32

#NoVideo

#NoAudio

#</Stream>

# Real with audio and video at 64 kbits

#<Stream test.rm>

#Feed feed1.ffm

#Format rm

#AudioBitRate 32

#VideoBitRate 128

#VideoFrameRate 25

#VideoGopSize 25

#NoAudio

#</Stream>

##################################################################

# A stream coming from a file: you only need to set the input

# filename and optionally a new format. Supported conversions:

#    AVI -> ASF

#<Stream file.rm>

#File "/usr/local/httpd/htdocs/tlive.rm"

#NoAudio

#</Stream>

#<Stream file.asf>

#File "/usr/local/httpd/htdocs/test.asf"

#NoAudio

#Author "Me"

#Copyright "Super MegaCorp"

#Title "Test stream from disk"

#Comment "Test comment"

#</Stream>

##################################################################

# RTSP examples

# You can access this stream with the RTSP URL:

#   rtsp://localhost:5454/test1-rtsp.mpg

# A non-standard RTSP redirector is also created. Its URL is:

#   http://localhost:8090/test1-rtsp.rtsp

#<Stream test1-rtsp.mpg>

#Format rtp

#File "/usr/local/httpd/htdocs/test1.mpg"

#</Stream>

# Transcode an incoming live feed to another live feed,

# using libx264 and video presets

#<Stream live.h264>

#Format rtp

#Feed feed1.ffm

#VideoCodec libx264

#VideoFrameRate 24

#VideoBitRate 100

#VideoSize 480x272

#AVPresetVideo default

#AVPresetVideo baseline

#AVOptionVideo flags +global_header

#AudioCodec libfaac

#AudioBitRate 32

#AudioChannels 2

#AudioSampleRate 22050

#AVOptionAudio flags +global_header

#</Stream>

##################################################################

# SDP/multicast examples

# If you want to send your stream in multicast, you must set the

# multicast address with MulticastAddress. The port and the TTL can

# also be set.

# An SDP file is automatically generated by ffserver by adding the

# 'sdp' extension to the stream name (here

# http://localhost:8090/test1-sdp.sdp). You should usually give this

# file to your player to play the stream.

# The 'NoLoop' option can be used to avoid looping when the stream is

# terminated.

#<Stream test1-sdp.mpg>

#Format rtp

#File "/usr/local/httpd/htdocs/test1.mpg"

#MulticastAddress 224.124.0.1

#MulticastPort 5000

#MulticastTTL 16

#NoLoop

#</Stream>

##################################################################

# Special streams

# Server status

Format status

# Only allow local people to get the status

ACL allow localhost

ACL allow 192.168.0.0 192.168.255.255

#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico

</Stream>

# Redirect index.html to the appropriate site

URL http://www.ffmpeg.org/

</Redirect>

[http_127.0.0.1_9999_feed1.ffm]: http://127.0.0.1:9999/feed1.ffm
[http_192.168.1.230_9999_test.flv]: http://192.168.1.230:9999/test.flv
[http_localhost_8090_feed1.ffm]: http://localhost:8090/feed1.ffm