FFMpeg處理RTMP流的總結
阿新 • • 發佈：2019-02-01
FFMpeg處理RTMP流有兩種方式:
一個是使用自帶的RTMP程式碼功能；
一個是使用第三方庫librtmp；
下面就這兩種方式的一些使用和差異做了個總結；
一、自帶RTMP程式碼功能
FFmpeg自帶的RTMP程式碼只支援RTMP協議，不支援rtmpt,rtmpe,rtmpte和rtmps協議；
命令列設定如下:
1. 將RTMP流原樣儲存成檔案
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec copy -vcodec copy -f flv -y test.flv
2. 將RTMP流轉碼儲存成檔案
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec ... -vcodec ... -f mp4 -y test.mp4
3. 將RTMP流轉碼後再以RTMP流的方式推送到RTMP流伺服器
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec ... -vcodec ... -f flv rtmp://10.2.11.111/live/newstream
NOTE:
FFMpeg自帶RTMP程式碼只支援RTMP流格式如:
rtmp://server:port/app/stream_name (eg: rtmp://192.168.1.11:80/live/test)
不支援RTMP流格式如:
rtmp://192.168.1.11:80/live/app/test
要想支援這種格式的RTMP流，就需要更專業和強大的每三方庫librtmp;
二、第三方庫librtmp
如何讓FFMpeg連結該庫可以參見文章:
http://blog.csdn.net/fireroll/article/details/8607955
這樣FFMpeg就可以支援rtmp://, rtmpt://, rtmpe://, rtmpte://,以及 rtmps://協議了。
連結了librtmp的FFMpeg接受一個字串的輸入方式，
如："rtmp://server:port/app/playpath/stream_name live=1 playpath=xxx ..."
NOTE：引號是必須的;
1. 儲存RTMP直播流原樣儲存成檔案：
# ./ffmpeg -i "rtmp://pub1.guoshi.com/live/newcetv1 live=1" -vcodec copy -acodec copy -y cetv1.flv
2. 將RTMP流轉碼後再以RTMP流的方式推送到RTMP流伺服器
# ./ffmpeg -i "rtmp://192.168.1.11:1935/live/app/teststream live=1" -acodec ... -vcodec ... -f flv rtmp://10.2.11.111/live/newstream
3. 用ffplay播放RTMP直播流：
ffplay "rtmp://pub1.guoshi.com/live/newcetv1 live=1"
4. 在使用FFMPEG類庫進行程式設計的時候，也是一樣的，
只需要將字串傳遞給avformat_open_input()就行了，形如：
ffplay "rtmp://pub1.guoshi.com/live/newcetv1 live=1"
char url[]="rtmp://live.hkstv.hk.lxdns.com/live/hks live=1";
avformat_open_input(&pFormatCtx,url,NULL,&avdic)
三、librtmp支援的引數：
http://rtmpdump.mplayerhq.hu/librtmp.3.html
1. NAME
librtmp RTMPDump Real-Time Messaging Protocol API
2. LIBRARY
RTMPDump RTMP (librtmp, -lrtmp)
3. SYNOPSIS
#include <librtmp/rtmp.h>
4. DESCRIPTION
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.
This API provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).
The basic RTMP specification has been published by Adobe but this API was reverse-engineered without use of the Adobe specification.
As such, it may deviate from any published specifications but it usually duplicates the actual behavior of the original Adobe clients.
The RTMPDump software package includes a basic client utility program in rtmpdump(1), some sample servers,
and a library used to provide programmatic access to the RTMP protocol. This man page gives an overview of the RTMP library routines.
These routines are found in the -lrtmp library. Many other routines are also available, but they are not documented yet.
The basic interaction is as follows. A session handle is created using RTMP_Alloc() and initialized using RTMP_Init().
All session parameters are provided using RTMP_SetupURL(). The network connection is established using RTMP_Connect(),
and then the RTMP session is established using RTMP_ConnectStream(). The stream is read using RTMP_Read().
A client can publish a stream by calling RTMP_EnableWrite() before the RTMP_Connect() call,
and then using RTMP_Write() after the session is established. While a stream is playing it may be paused and unpaused
using RTMP_Pause(). The stream playback position can be moved using RTMP_Seek(). When RTMP_Read() returns 0 bytes,
the stream is complete and may be closed using RTMP_Close(). The session handle is freed using RTMP_Free().
All data is transferred using FLV format. The basic session requires an RTMP URL. The RTMP URL format is of the form
rtmp[t][e|s]://hostname[:port][/app[/playpath]]
Plain rtmp, as well as tunneled and encrypted sessions are supported.
Additional options may be specified by appending space-separated key=value pairs to the URL.
Special characters in values may need to be escaped to prevent misinterpretation by the option parser.
The escape encoding uses a backslash followed by two hexadecimal digits representing the ASCII value of the character.
E.g., spaces must be escaped as \20 and backslashes must be escaped as \5c.
5. OPTIONS
5.1 Network Parameters
These options define how to connect to the media server.
socks=host:port
Use the specified SOCKS4 proxy.
5.2 Connection Parameters
These options define the content of the RTMP Connect request packet.
If correct values are not provided, the media server will reject the connection attempt.
app=name
Name of application to connect to on the RTMP server. Overrides the app in the RTMP URL.
Sometimes the librtmp URL parser cannot determine the app name automatically,
so it must be given explicitly using this option.
tcUrl=url
URL of the target stream. Defaults to rtmp[t][e|s]://host[:port]/app.
pageUrl=url
URL of the web page in which the media was embedded. By default no value will be sent.
swfUrl=url
URL of the SWF player for the media. By default no value will be sent.
flashVer=version
Version of the Flash plugin used to run the SWF player. The default is "LNX 10,0,32,18".
conn=type:data
Append arbitrary AMF data to the Connect message.
The type must be B for Boolean, N for number, S for string, O for object, or Z for null.
For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively.
Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively.
Data items in subobjects may be named, by prefixing the type with 'N' and specifying the name before the value,
e.g. NB:myFlag:1. This option may be used multiple times to construct arbitrary AMF sequences. E.g.
conn=B:1 conn=S:authMe conn=O:1 conn=NN:code:1.23 conn=NS:flag:ok conn=O:0
5.3 Session Parameters
These options take effect after the Connect request has succeeded.
playpath=path
Overrides the playpath parsed from the RTMP URL.
Sometimes the rtmpdump URL parser cannot determine the correct playpath automatically,
so it must be given explicitly using this option.
playlist=0|1
If the value is 1 or TRUE, issue a set_playlist command before sending the play command.
The playlist will just contain the current playpath. If the value is 0 or FALSE, the set_playlist command will not be sent.
The default is FALSE.
live=0|1
Specify that the media is a live stream. No resuming or seeking in live streams is possible.
subscribe=path
Name of live stream to subscribe to. Defaults to playpath.
start=num
Start at num seconds into the stream. Not valid for live streams.
stop=num
Stop at num seconds into the stream.
buffer=num
Set buffer time to num milliseconds. The default is 30000.
timeout=num
Timeout the session after num seconds without receiving any data from the server. The default is 120.
5.4 Security Parameters
These options handle additional authentication requests from the server.
token=key
Key for SecureToken response, used if the server requires SecureToken authentication.
jtv=JSON
JSON token used by legacy Justin.tv servers. Invokes NetStream.Authenticate.UsherToken
swfVfy=0|1
If the value is 1 or TRUE, the SWF player is retrieved from the specified swfUrl for performing SWF Verification.
The SWF hash and size (used in the verification step) are computed automatically.
Also the SWF information is cached in a .swfinfo file in the user's home directory,
so that it doesn't need to be retrieved and recalculated every time. The .swfinfo file records the SWF URL,
the time it was fetched, the modification timestamp of the SWF file, its size, and its hash.
By default, the cached info will be used for 30 days before re-checking.
swfAge=days
Specify how many days to use the cached SWF info before re-checking.
Use 0 to always check the SWF URL. Note that if the check shows that the SWF file has the same modification timestamp as before,
it will not be retrieved again.
5.5 EXAMPLES
An example character string suitable for use with RTMP_SetupURL():
"rtmp://flashserver:1935/ondemand/thefile swfUrl=http://flashserver/player.swf swfVfy=1"
一個是使用自帶的RTMP程式碼功能；
一個是使用第三方庫librtmp；
下面就這兩種方式的一些使用和差異做了個總結；
一、自帶RTMP程式碼功能
FFmpeg自帶的RTMP程式碼只支援RTMP協議，不支援rtmpt,rtmpe,rtmpte和rtmps協議；
命令列設定如下:
1. 將RTMP流原樣儲存成檔案
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec copy -vcodec copy -f flv -y test.flv
2. 將RTMP流轉碼儲存成檔案
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec ... -vcodec ... -f mp4 -y test.mp4
3. 將RTMP流轉碼後再以RTMP流的方式推送到RTMP流伺服器
# ./ffmpeg -i rtmp://192.168.1.11:1935/live/teststream -acodec ... -vcodec ... -f flv rtmp://10.2.11.111/live/newstream
NOTE:
FFMpeg自帶RTMP程式碼只支援RTMP流格式如:
rtmp://server:port/app/stream_name (eg: rtmp://192.168.1.11:80/live/test)
不支援RTMP流格式如:
rtmp://192.168.1.11:80/live/app/test
要想支援這種格式的RTMP流，就需要更專業和強大的每三方庫librtmp;
二、第三方庫librtmp
如何讓FFMpeg連結該庫可以參見文章:
http://blog.csdn.net/fireroll/article/details/8607955
這樣FFMpeg就可以支援rtmp://, rtmpt://, rtmpe://, rtmpte://,以及 rtmps://協議了。
連結了librtmp的FFMpeg接受一個字串的輸入方式，
如："rtmp://server:port/app/playpath/stream_name live=1 playpath=xxx ..."
NOTE：引號是必須的;
1. 儲存RTMP直播流原樣儲存成檔案：
# ./ffmpeg -i "rtmp://pub1.guoshi.com/live/newcetv1 live=1" -vcodec copy -acodec copy -y cetv1.flv
2. 將RTMP流轉碼後再以RTMP流的方式推送到RTMP流伺服器
# ./ffmpeg -i "rtmp://192.168.1.11:1935/live/app/teststream live=1" -acodec ... -vcodec ... -f flv rtmp://10.2.11.111/live/newstream
3. 用ffplay播放RTMP直播流：
ffplay "rtmp://pub1.guoshi.com/live/newcetv1 live=1"
4. 在使用FFMPEG類庫進行程式設計的時候，也是一樣的，
只需要將字串傳遞給avformat_open_input()就行了，形如：
ffplay "rtmp://pub1.guoshi.com/live/newcetv1 live=1"
char url[]="rtmp://live.hkstv.hk.lxdns.com/live/hks live=1";
avformat_open_input(&pFormatCtx,url,NULL,&avdic)
三、librtmp支援的引數：
http://rtmpdump.mplayerhq.hu/librtmp.3.html
1. NAME
librtmp RTMPDump Real-Time Messaging Protocol API
2. LIBRARY
RTMPDump RTMP (librtmp, -lrtmp)
3. SYNOPSIS
#include <librtmp/rtmp.h>
4. DESCRIPTION
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.
This API provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).
The basic RTMP specification has been published by Adobe but this API was reverse-engineered without use of the Adobe specification.
As such, it may deviate from any published specifications but it usually duplicates the actual behavior of the original Adobe clients.
The RTMPDump software package includes a basic client utility program in rtmpdump(1), some sample servers,
and a library used to provide programmatic access to the RTMP protocol. This man page gives an overview of the RTMP library routines.
These routines are found in the -lrtmp library. Many other routines are also available, but they are not documented yet.
The basic interaction is as follows. A session handle is created using RTMP_Alloc() and initialized using RTMP_Init().
All session parameters are provided using RTMP_SetupURL(). The network connection is established using RTMP_Connect(),
and then the RTMP session is established using RTMP_ConnectStream(). The stream is read using RTMP_Read().
A client can publish a stream by calling RTMP_EnableWrite() before the RTMP_Connect() call,
and then using RTMP_Write() after the session is established. While a stream is playing it may be paused and unpaused
using RTMP_Pause(). The stream playback position can be moved using RTMP_Seek(). When RTMP_Read() returns 0 bytes,
the stream is complete and may be closed using RTMP_Close(). The session handle is freed using RTMP_Free().
All data is transferred using FLV format. The basic session requires an RTMP URL. The RTMP URL format is of the form
rtmp[t][e|s]://hostname[:port][/app[/playpath]]
Plain rtmp, as well as tunneled and encrypted sessions are supported.
Additional options may be specified by appending space-separated key=value pairs to the URL.
Special characters in values may need to be escaped to prevent misinterpretation by the option parser.
The escape encoding uses a backslash followed by two hexadecimal digits representing the ASCII value of the character.
E.g., spaces must be escaped as \20 and backslashes must be escaped as \5c.
5. OPTIONS
5.1 Network Parameters
These options define how to connect to the media server.
socks=host:port
Use the specified SOCKS4 proxy.
5.2 Connection Parameters
These options define the content of the RTMP Connect request packet.
If correct values are not provided, the media server will reject the connection attempt.
app=name
Name of application to connect to on the RTMP server. Overrides the app in the RTMP URL.
Sometimes the librtmp URL parser cannot determine the app name automatically,
so it must be given explicitly using this option.
tcUrl=url
URL of the target stream. Defaults to rtmp[t][e|s]://host[:port]/app.
pageUrl=url
URL of the web page in which the media was embedded. By default no value will be sent.
swfUrl=url
URL of the SWF player for the media. By default no value will be sent.
flashVer=version
Version of the Flash plugin used to run the SWF player. The default is "LNX 10,0,32,18".
conn=type:data
Append arbitrary AMF data to the Connect message.
The type must be B for Boolean, N for number, S for string, O for object, or Z for null.
For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively.
Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively.
Data items in subobjects may be named, by prefixing the type with 'N' and specifying the name before the value,
e.g. NB:myFlag:1. This option may be used multiple times to construct arbitrary AMF sequences. E.g.
conn=B:1 conn=S:authMe conn=O:1 conn=NN:code:1.23 conn=NS:flag:ok conn=O:0
5.3 Session Parameters
These options take effect after the Connect request has succeeded.
playpath=path
Overrides the playpath parsed from the RTMP URL.
Sometimes the rtmpdump URL parser cannot determine the correct playpath automatically,
so it must be given explicitly using this option.
playlist=0|1
If the value is 1 or TRUE, issue a set_playlist command before sending the play command.
The playlist will just contain the current playpath. If the value is 0 or FALSE, the set_playlist command will not be sent.
The default is FALSE.
live=0|1
Specify that the media is a live stream. No resuming or seeking in live streams is possible.
subscribe=path
Name of live stream to subscribe to. Defaults to playpath.
start=num
Start at num seconds into the stream. Not valid for live streams.
stop=num
Stop at num seconds into the stream.
buffer=num
Set buffer time to num milliseconds. The default is 30000.
timeout=num
Timeout the session after num seconds without receiving any data from the server. The default is 120.
5.4 Security Parameters
These options handle additional authentication requests from the server.
token=key
Key for SecureToken response, used if the server requires SecureToken authentication.
jtv=JSON
JSON token used by legacy Justin.tv servers. Invokes NetStream.Authenticate.UsherToken
swfVfy=0|1
If the value is 1 or TRUE, the SWF player is retrieved from the specified swfUrl for performing SWF Verification.
The SWF hash and size (used in the verification step) are computed automatically.
Also the SWF information is cached in a .swfinfo file in the user's home directory,
so that it doesn't need to be retrieved and recalculated every time. The .swfinfo file records the SWF URL,
the time it was fetched, the modification timestamp of the SWF file, its size, and its hash.
By default, the cached info will be used for 30 days before re-checking.
swfAge=days
Specify how many days to use the cached SWF info before re-checking.
Use 0 to always check the SWF URL. Note that if the check shows that the SWF file has the same modification timestamp as before,
it will not be retrieved again.
5.5 EXAMPLES
An example character string suitable for use with RTMP_SetupURL():
"rtmp://flashserver:1935/ondemand/thefile swfUrl=http://flashserver/player.swf swfVfy=1"