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 |