 Chromium Code Reviews
 Chromium Code Reviews Issue 1315903004:
  ABANDONED: Remove the default receive channel in WVoE.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@mediacontroller
    
  
    Issue 1315903004:
  ABANDONED: Remove the default receive channel in WVoE.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@mediacontroller| Index: talk/media/webrtc/webrtcvoiceengine.h | 
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h | 
| index 7950024712b1ce35aeed67ab4306360f1be06424..5a7952dbfaacb02bfcdf71bd72d5c7cef49b4fc4 100644 | 
| --- a/talk/media/webrtc/webrtcvoiceengine.h | 
| +++ b/talk/media/webrtc/webrtcvoiceengine.h | 
| @@ -246,6 +246,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, | 
| bool SetSendRtpHeaderExtensions( | 
| const std::vector<RtpHeaderExtension>& extensions); | 
| bool SetOptions(const AudioOptions& options); | 
| + bool SetRecvOptions(int channel_id); | 
| bool SetMaxSendBandwidth(int bps); | 
| bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 
| bool SetRecvRtpHeaderExtensions( | 
| @@ -260,9 +261,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, | 
| const std::vector<AudioCodec>& all_codecs, | 
| webrtc::CodecInst* send_codec); | 
| bool EnableRtcp(int channel); | 
| - bool ResetRecvCodecs(int channel); | 
| bool SetPlayout(int channel, bool playout); | 
| - static uint32 ParseSsrc(const void* data, size_t len, bool rtcp); | 
| static Error WebRtcErrorToChannelError(int err_code); | 
| class WebRtcVoiceChannelRenderer; | 
| @@ -283,9 +282,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, | 
| void ConfigureSendChannel(int channel); | 
| bool ConfigureRecvChannel(int channel); | 
| bool DeleteChannel(int channel); | 
| - bool InConferenceMode() const { | 
| - return options_.conference_mode.GetWithDefaultIfUnset(false); | 
| - } | 
| bool IsDefaultChannel(int channel_id) const { | 
| return channel_id == voe_channel(); | 
| } | 
| @@ -325,16 +321,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, | 
| SendFlags send_; | 
| webrtc::Call* const call_; | 
| + uint32_t default_recv_ssrc_; | 
| + int default_recv_channel_id_; | 
| 
pthatcher1
2015/10/02 02:33:31
Why do we store the default_recv_channel_id_ when
 
the sun
2015/10/02 11:34:20
You're right; I was using it as a flag anyway.
 | 
| + | 
| // send_channels_ contains the channels which are being used for sending. | 
| // When the default channel (voe_channel) is used for sending, it is | 
| // contained in send_channels_, otherwise not. | 
| ChannelMap send_channels_; | 
| std::vector<RtpHeaderExtension> send_extensions_; | 
| - uint32 default_receive_ssrc_; | 
| - // Note the default channel (voe_channel()) can reside in both | 
| - // receive_channels_ and send_channels_ in non-conference mode and in that | 
| - // case it will only be there if a non-zero default_receive_ssrc_ is set. | 
| - ChannelMap receive_channels_; // for multiple sources | 
| + ChannelMap receive_channels_; | 
| std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_; | 
| std::map<uint32, StreamParams> receive_stream_params_; | 
| // receive_channels_ can be read from WebRtc callback thread. Access from | 
| @@ -342,10 +337,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, | 
| // Reads on the worker thread are ok. | 
| std::vector<RtpHeaderExtension> receive_extensions_; | 
| std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 
| - | 
| - // Do not lock this on the VoE media processor thread; potential for deadlock | 
| - // exists. | 
| - mutable rtc::CriticalSection receive_channels_cs_; | 
| }; | 
| } // namespace cricket |