Index: talk/media/webrtc/webrtcvoiceengine.h |
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
index cd792bcb435843fe2320cbc7bd3d42fa658220a2..a8f3ec85115e388cf416d7cc24097e593673807e 100644 |
--- a/talk/media/webrtc/webrtcvoiceengine.h |
+++ b/talk/media/webrtc/webrtcvoiceengine.h |
@@ -115,9 +115,6 @@ class WebRtcVoiceEngine |
// Stops recording the RtcEventLog. |
void StopRtcEventLog(); |
- // Create a VoiceEngine Channel. |
- int CreateMediaVoiceChannel(); |
- |
private: |
void Construct(); |
void ConstructCodecs(); |
@@ -143,7 +140,7 @@ class WebRtcVoiceEngine |
void StartAecDump(const std::string& filename); |
void StopAecDump(); |
- int CreateVoiceChannel(VoEWrapper* voe); |
+ int CreateVoEChannel(); |
static const int kDefaultLogSeverity = rtc::LS_WARNING; |
@@ -187,8 +184,6 @@ 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; |
@@ -267,7 +262,6 @@ 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); |
@@ -280,18 +274,13 @@ 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); |
} |
@@ -315,7 +304,6 @@ 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_; |
@@ -335,6 +323,9 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
int64_t default_recv_ssrc_ = -1; |
// Volume for unsignalled stream, which may be set before the stream exists. |
double default_recv_volume_ = 1.0; |
+ // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled |
+ // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
+ uint32_t receiver_reports_ssrc_ = 1; |
// send_channels_ contains the channels which are being used for sending. |
// When the default channel (default_send_channel_id) is used for sending, it |