Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.h |
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
| index 4250741e05ce30a8c6488bcaefec86bcdb9e13de..9dc728c4254f3a0892b35c8c2a5d1d5d9050c73b 100644 |
| --- a/talk/media/webrtc/webrtcvoiceengine.h |
| +++ b/talk/media/webrtc/webrtcvoiceengine.h |
| @@ -259,9 +259,7 @@ 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_t ParseSsrc(const void* data, size_t len, bool rtcp); |
| static Error WebRtcErrorToChannelError(int err_code); |
| class WebRtcVoiceChannelRenderer; |
| @@ -282,12 +280,12 @@ 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 == default_send_channel_id_; |
| } |
| + bool IsDefaultRecvStream(uint32_t ssrc) { |
| + return default_recv_ssrc_ == static_cast<int64_t>(ssrc); |
| + } |
| bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); |
| bool SetSendBitrateInternal(int bps); |
| @@ -324,16 +322,16 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| SendFlags send_; |
| webrtc::Call* const call_; |
| + // SSRC of unsignalled receive stream, or -1 if there isn't one. |
| + int64_t default_recv_ssrc_; |
| + double default_recv_volume_; |
|
pthatcher1
2015/10/13 05:38:46
Can you leave a comment about when this is set and
the sun
2015/10/13 10:03:28
Done.
|
| + |
| // send_channels_ contains the channels which are being used for sending. |
| // When the default channel (default_send_channel_id) is used for sending, it |
| // is contained in send_channels_, otherwise not. |
| ChannelMap send_channels_; |
| std::vector<RtpHeaderExtension> send_extensions_; |
| - uint32_t default_receive_ssrc_; |
| - // Note the default channel (default_send_channel_id()) 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_t, webrtc::AudioReceiveStream*> receive_streams_; |
| std::map<uint32_t, StreamParams> receive_stream_params_; |
| // receive_channels_ can be read from WebRtc callback thread. Access from |
| @@ -341,10 +339,6 @@ 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 |