diff options
Diffstat (limited to 'media/webrtc/webrtcvoiceengine.cc')
| -rw-r--r-- | media/webrtc/webrtcvoiceengine.cc | 24 |
1 files changed, 21 insertions, 3 deletions
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc index 745a1e0..2aa6b8c 100644 --- a/media/webrtc/webrtcvoiceengine.cc +++ b/media/webrtc/webrtcvoiceengine.cc @@ -1433,6 +1433,22 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm, return true; } +bool WebRtcVoiceEngine::StartAecDump(FILE* file) { +#ifdef USE_WEBRTC_DEV_BRANCH + StopAecDump(); + if (voe_wrapper_->processing()->StartDebugRecording(file) != + webrtc::AudioProcessing::kNoError) { + LOG_RTCERR1(StartDebugRecording, "FILE*"); + fclose(file); + return false; + } + is_dumping_aec_ = true; + return true; +#else + return false; +#endif +} + bool WebRtcVoiceEngine::RegisterProcessor( uint32 ssrc, VoiceProcessor* voice_processor, @@ -1590,7 +1606,7 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { // Start dumping AEC when we are not dumping. if (voe_wrapper_->processing()->StartDebugRecording( filename.c_str()) != webrtc::AudioProcessing::kNoError) { - LOG_RTCERR0(StartDebugRecording); + LOG_RTCERR1(StartDebugRecording, filename.c_str()); } else { is_dumping_aec_ = true; } @@ -2821,7 +2837,8 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event, return true; } -void WebRtcVoiceMediaChannel::OnPacketReceived(talk_base::Buffer* packet) { +void WebRtcVoiceMediaChannel::OnPacketReceived( + talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) { // Pick which channel to send this packet to. If this packet doesn't match // any multiplexed streams, just send it to the default channel. Otherwise, // send it to the specific decoder instance for that stream. @@ -2854,7 +2871,8 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(talk_base::Buffer* packet) { static_cast<unsigned int>(packet->length())); } -void WebRtcVoiceMediaChannel::OnRtcpReceived(talk_base::Buffer* packet) { +void WebRtcVoiceMediaChannel::OnRtcpReceived( + talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) { // Sending channels need all RTCP packets with feedback information. // Even sender reports can contain attached report blocks. // Receiving channels need sender reports in order to create |