lenn/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-11-23 19:34:05 +08:00
Code Issues Packages Projects Releases Wiki Activity

AI: WebRTC: Support G.711 (PCMU/PCMA) audio codec for WebRTC. v7.0.124 (#4075) (#4568)

Browse Source 
This PR adds G.711 (PCMU/PCMA) audio codec support for WebRTC in SRS,
enabling relay-only streaming of G.711 audio between WebRTC clients via
WHIP/WHEP. G.711 is a widely-used, royalty-free audio codec with
excellent compatibility across VoIP systems, IP cameras, and legacy
telephony equipment.

Fixes #4075

Many IP cameras, VoIP systems, and IoT devices use G.711 (PCMU/PCMA) as
their default audio codec. Previously, SRS only supported Opus for
WebRTC audio, requiring transcoding or rejecting G.711 streams entirely.
This PR enables direct relay of G.711 audio streams in WebRTC, similar
to how VP9/AV1 video codecs are supported.

Enhanced WHIP/WHEP players with URL-based codec selection:
```
# Audio codec only
http://localhost:8080/players/whip.html?acodec=pcmu
http://localhost:8080/players/whip.html?acodec=pcma

# Video + audio codecs
http://localhost:8080/players/whip.html?vcodec=vp9&acodec=pcmu
http://localhost:8080/players/whep.html?vcodec=h264&acodec=pcma

# Backward compatible (codec = vcodec)
http://localhost:8080/players/whip.html?codec=vp9
```

Testing

```bash
# Build and run unit tests
cd trunk
make utest -j && ./objs/srs_utest

# Test with WHIP player
# 1. Start SRS server
./objs/srs -c conf/rtc.conf

# 2. Open WHIP publisher with PCMU audio
http://localhost:8080/players/whip.html?acodec=pcmu

# 3. Open WHEP player to receive stream
http://localhost:8080/players/whep.html
```

## Related Issues

- Fixes #4075 - WebRTC G.711A Audio Codec Support
- Related to #4548 - VP9 codec support (similar relay-only pattern)
This commit is contained in:
OSSRS-AI
2025-11-09 12:08:03 -05:00
committed by GitHub
parent 7fcd406a63
commit bfb91f9b82
20 changed files with 463 additions and 43 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
trunk/research/players/js/srs.page.js
View File

@@ -24,7 +24,7 @@ function update_nav() {
$("#nav_vlc").attr("href", "vlc.html" + window.location.search);
}
// Special extra params, such as auth_key.
// Special extra params, such as auth_key, codec, vcodec, acodec.
function user_extra_params(query, params, rtc) {
var queries = params || [];
@@ -124,6 +124,9 @@ function build_default_whip_whep_url(query, apiPath) {
console.log('?api=x to overwrite WebRTC API(1985).');
console.log('?schema=http|https to overwrite WebRTC API protocol.');
console.log(`?path=xxx to overwrite default ${apiPath}`);
console.log('?codec=xxx to specify video codec (alias for vcodec, e.g., h264, vp9, av1)');
console.log('?vcodec=xxx to specify video codec (e.g., h264, vp9, av1)');
console.log('?acodec=xxx to specify audio codec (e.g., opus, pcmu, pcma)');
var server = (!query.server)? window.location.hostname:query.server;
var vhost = (!query.vhost)? window.location.hostname:query.vhost;

59
trunk/research/players/js/srs.sdk.js
View File

@@ -41,11 +41,15 @@ function SrsRtcWhipWhepAsync() {
// camera: boolean, whether capture video from camera, default to true.
// screen: boolean, whether capture video from screen, default to false.
// audio: boolean, whether play audio, default to true.
// vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
// acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
self.publish = async function (url, options) {
if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
const hasAudio = options?.audio ?? true;
const useCamera = options?.camera ?? true;
const useScreen = options?.screen ?? false;
const vcodec = options?.vcodec;
const acodec = options?.acodec;
if (!hasAudio && !useCamera && !useScreen) throw new Error(`The camera, screen and audio can't be false at the same time`);
@@ -91,6 +95,13 @@ function SrsRtcWhipWhepAsync() {
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
// Filter codecs if specified
if (vcodec || acodec) {
offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
}
const answer = await new Promise(function (resolve, reject) {
console.log(`Generated offer: ${offer.sdp}`);
@@ -119,15 +130,26 @@ function SrsRtcWhipWhepAsync() {
// @options The options to control playing, supports:
// videoOnly: boolean, whether only play video, default to false.
// audioOnly: boolean, whether only play audio, default to false.
// vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
// acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
self.play = async function(url, options) {
if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
const vcodec = options?.vcodec;
const acodec = options?.acodec;
if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"});
if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
// Filter codecs if specified
if (vcodec || acodec) {
offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
}
const answer = await new Promise(function(resolve, reject) {
console.log(`Generated offer: ${offer.sdp}`);
@@ -199,6 +221,43 @@ function SrsRtcWhipWhepAsync() {
simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
};
},
filterCodec: (sdp, vcodec, acodec) => {
// Filter video codec if specified
if (vcodec) {
const vcodecUpper = vcodec.toUpperCase();
sdp = sdp.split('\n').filter(line => {
// Keep all non-video lines
if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
!line.startsWith('a=fmtp:')) {
return true;
}
// For video codec lines, only keep the specified codec
if (line.includes('video/')) {
return line.toUpperCase().includes(vcodecUpper);
}
return true;
}).join('\n');
}
// Filter audio codec if specified
if (acodec) {
const acodecUpper = acodec.toUpperCase();
sdp = sdp.split('\n').filter(line => {
// Keep all non-audio lines
if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
!line.startsWith('a=fmtp:')) {
return true;
}
// For audio codec lines, only keep the specified codec
if (line.includes('audio/')) {
return line.toUpperCase().includes(acodecUpper);
}
return true;
}).join('\n');
}
return sdp;
},
};
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack

8
trunk/research/players/whep.html
View File

@@ -125,9 +125,17 @@ $(function(){
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
var query = parse_query_string();
// Support codec parameters: codec (alias for vcodec), vcodec, acodec
var vcodec = query.vcodec || query.codec;
var acodec = query.acodec;
sdk.play(url, {
videoOnly: $('#ch_videoonly').prop('checked'),
audioOnly: $('#ch_audioonly').prop('checked'),
vcodec: vcodec,
acodec: acodec
}).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);

10
trunk/research/players/whip.html
View File

@@ -132,10 +132,18 @@ $(function(){
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
var query = parse_query_string();
// Support codec parameters: codec (alias for vcodec), vcodec, acodec
var vcodec = query.vcodec || query.codec;
var acodec = query.acodec;
sdk.publish(url, {
camera: $('#ra_camera').prop('checked'),
screen: $('#ra_screen').prop('checked'),
audio: $('#ch_audio').prop('checked')
audio: $('#ch_audio').prop('checked'),
vcodec: vcodec,
acodec: acodec
}).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);