Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.h |
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
| index 5121c08b7950c6e68508a8cef0dcd705ca09f4ea..0abf17b55fe821064d3d2ce1452c643cd59feaf1 100644 |
| --- a/talk/media/webrtc/webrtcvoiceengine.h |
| +++ b/talk/media/webrtc/webrtcvoiceengine.h |
| @@ -108,9 +108,6 @@ class WebRtcVoiceEngine |
| // Starts AEC dump using existing file. |
| bool StartAecDump(rtc::PlatformFile file); |
| - // Create a VoiceEngine Channel. |
| - int CreateMediaVoiceChannel(); |
| - |
| private: |
| void Construct(); |
| void ConstructCodecs(); |
| @@ -136,7 +133,7 @@ class WebRtcVoiceEngine |
| void StartAecDump(const std::string& filename); |
| void StopAecDump(); |
| - int CreateVoiceChannel(VoEWrapper* voe); |
| + int CreateVoEChannel(); |
| static const int kDefaultLogSeverity = rtc::LS_WARNING; |
| @@ -180,8 +177,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| webrtc::Call* call); |
| ~WebRtcVoiceMediaChannel() override; |
| - int default_send_channel_id() const { return default_send_channel_id_; } |
| - bool valid() const { return default_send_channel_id_ != -1; } |
| const AudioOptions& options() const { return options_; } |
| bool SetSendParameters(const AudioSendParameters& params) override; |
| @@ -258,7 +253,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| bool GetRedSendCodec(const AudioCodec& red_codec, |
| const std::vector<AudioCodec>& all_codecs, |
| webrtc::CodecInst* send_codec); |
| - bool EnableRtcp(int channel); |
| bool SetPlayout(int channel, bool playout); |
| static Error WebRtcErrorToChannelError(int err_code); |
| @@ -271,18 +265,14 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| unsigned char); |
| void SetNack(int channel, bool nack_enabled); |
| - void SetNack(const ChannelMap& channels, bool nack_enabled); |
| bool SetSendCodec(const webrtc::CodecInst& send_codec); |
| bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); |
| bool ChangePlayout(bool playout); |
| bool ChangeSend(SendFlags send); |
| bool ChangeSend(int channel, SendFlags send); |
| - void ConfigureSendChannel(int channel); |
| bool ConfigureRecvChannel(int channel); |
| + int CreateVoEChannel(); |
| bool DeleteChannel(int channel); |
| - 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); |
| } |
| @@ -306,7 +296,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| rtc::ThreadChecker thread_checker_; |
| WebRtcVoiceEngine* const engine_; |
| - const int default_send_channel_id_; |
| std::vector<AudioCodec> recv_codecs_; |
| std::vector<AudioCodec> send_codecs_; |
| rtc::scoped_ptr<webrtc::CodecInst> send_codec_; |
| @@ -327,6 +316,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| // Volume for unsignalled stream, which may be set before the stream exists. |
| double default_recv_volume_ = 1.0; |
| + uint32_t first_send_ssrc_ = -1; |
|
pthatcher1
2015/10/15 05:15:04
Can you comment more on what this means and when i
the sun
2015/10/15 12:07:12
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. |