| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 7f0e89c13e7c5fa2ec5dbe45e30423c8d6857f71..66e82f1d8f4b4412d3aaf7ad66bab8cd0e2d4943 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -251,6 +251,7 @@ 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(
|
| @@ -265,9 +266,7 @@ 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;
|
| @@ -288,9 +287,6 @@ 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();
|
| }
|
| @@ -330,16 +326,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| SendFlags send_;
|
| webrtc::Call* const call_;
|
|
|
| + uint32_t default_recv_ssrc_;
|
| + int default_recv_channel_id_;
|
| +
|
| // 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
|
| @@ -347,10 +342,6 @@ 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
|
|
|