Chromium Code Reviews| 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 |