Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index b39be5b2b253e4f853cb731176cf9bbe2a833a66..718883c90b8100b2d3121ad7e61616e382596c58 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -843,6 +843,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
webrtc::AudioTransport* voe_audio_transport, |
uint32_t ssrc, |
const std::string& c_name, |
+ const std::string track_id, |
const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>& |
send_codec_spec, |
const std::vector<webrtc::RtpExtension>& extensions, |
@@ -869,6 +870,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
config_.rtp.extensions = extensions; |
config_.audio_network_adaptor_config = audio_network_adaptor_config; |
config_.encoder_factory = encoder_factory; |
+ config_.track_id = track_id; |
rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc); |
if (send_codec_spec) { |
@@ -1870,7 +1872,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { |
rtc::Optional<std::string> audio_network_adaptor_config = |
GetAudioNetworkAdaptorConfig(options_); |
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( |
- channel, audio_transport, ssrc, sp.cname, send_codec_spec_, |
+ channel, audio_transport, ssrc, sp.cname, sp.id, send_codec_spec_, |
send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config, |
call_, this, engine()->encoder_factory_); |
send_streams_.insert(std::make_pair(ssrc, stream)); |