| Index: webrtc/voice_engine/channel.cc
|
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
|
| index fb29dfabab35ee5dbcf3d481de0449d65f8060e2..8362f302b4cb49e894dc9c2c0f325b56737bcc0f 100644
|
| --- a/webrtc/voice_engine/channel.cc
|
| +++ b/webrtc/voice_engine/channel.cc
|
| @@ -217,14 +217,10 @@ Channel::OnRxVadDetected(int vadDecision)
|
| }
|
|
|
| int
|
| -Channel::SendPacket(int channel, const void *data, size_t len)
|
| +Channel::SendPacket(const void *data, size_t len)
|
| {
|
| - channel = VoEChannelId(channel);
|
| - assert(channel == _channelId);
|
| -
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::SendPacket(channel=%d, len=%" PRIuS ")", channel,
|
| - len);
|
| + "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
|
|
|
| CriticalSectionScoped cs(&_callbackCritSect);
|
|
|
| @@ -239,8 +235,7 @@ Channel::SendPacket(int channel, const void *data, size_t len)
|
| uint8_t* bufferToSendPtr = (uint8_t*)data;
|
| size_t bufferLength = len;
|
|
|
| - int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
|
| - bufferLength);
|
| + int n = _transportPtr->SendPacket(bufferToSendPtr, bufferLength);
|
| if (n < 0) {
|
| std::string transport_name =
|
| _externalTransport ? "external transport" : "WebRtc sockets";
|
| @@ -254,14 +249,10 @@ Channel::SendPacket(int channel, const void *data, size_t len)
|
| }
|
|
|
| int
|
| -Channel::SendRTCPPacket(int channel, const void *data, size_t len)
|
| +Channel::SendRTCPPacket(const void *data, size_t len)
|
| {
|
| - channel = VoEChannelId(channel);
|
| - assert(channel == _channelId);
|
| -
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::SendRTCPPacket(channel=%d, len=%" PRIuS ")", channel,
|
| - len);
|
| + "Channel::SendRTCPPacket(len=%" PRIuS ")", len);
|
|
|
| CriticalSectionScoped cs(&_callbackCritSect);
|
| if (_transportPtr == NULL)
|
| @@ -276,9 +267,7 @@ Channel::SendRTCPPacket(int channel, const void *data, size_t len)
|
| uint8_t* bufferToSendPtr = (uint8_t*)data;
|
| size_t bufferLength = len;
|
|
|
| - int n = _transportPtr->SendRTCPPacket(channel,
|
| - bufferToSendPtr,
|
| - bufferLength);
|
| + int n = _transportPtr->SendRTCPPacket(bufferToSendPtr, bufferLength);
|
| if (n < 0) {
|
| std::string transport_name =
|
| _externalTransport ? "external transport" : "WebRtc sockets";
|
| @@ -291,15 +280,12 @@ Channel::SendRTCPPacket(int channel, const void *data, size_t len)
|
| return n;
|
| }
|
|
|
| -void
|
| -Channel::OnPlayTelephoneEvent(int32_t id,
|
| - uint8_t event,
|
| - uint16_t lengthMs,
|
| - uint8_t volume)
|
| -{
|
| +void Channel::OnPlayTelephoneEvent(uint8_t event,
|
| + uint16_t lengthMs,
|
| + uint8_t volume) {
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
|
| - " volume=%u)", id, event, lengthMs, volume);
|
| + "Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
|
| + " volume=%u)", event, lengthMs, volume);
|
|
|
| if (!_playOutbandDtmfEvent || (event > 15))
|
| {
|
| @@ -316,40 +302,31 @@ Channel::OnPlayTelephoneEvent(int32_t id,
|
| }
|
|
|
| void
|
| -Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
|
| +Channel::OnIncomingSSRCChanged(uint32_t ssrc)
|
| {
|
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
|
| - id, ssrc);
|
| + "Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
|
|
|
| // Update ssrc so that NTP for AV sync can be updated.
|
| _rtpRtcpModule->SetRemoteSSRC(ssrc);
|
| }
|
|
|
| -void Channel::OnIncomingCSRCChanged(int32_t id,
|
| - uint32_t CSRC,
|
| - bool added)
|
| -{
|
| - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
|
| - id, CSRC, added);
|
| +void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
|
| + WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
|
| + "Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
|
| + added);
|
| }
|
|
|
| -int32_t
|
| -Channel::OnInitializeDecoder(
|
| - int32_t id,
|
| +int32_t Channel::OnInitializeDecoder(
|
| int8_t payloadType,
|
| const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
| int frequency,
|
| uint8_t channels,
|
| - uint32_t rate)
|
| -{
|
| + uint32_t rate) {
|
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
|
| + "Channel::OnInitializeDecoder(payloadType=%d, "
|
| "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
|
| - id, payloadType, payloadName, frequency, channels, rate);
|
| -
|
| - assert(VoEChannelId(id) == _channelId);
|
| + payloadType, payloadName, frequency, channels, rate);
|
|
|
| CodecInst receiveCodec = {0};
|
| CodecInst dummyCodec = {0};
|
| @@ -725,8 +702,7 @@ Channel::Channel(int32_t channelId,
|
| rtp_receive_statistics_(
|
| ReceiveStatistics::Create(Clock::GetRealTimeClock())),
|
| rtp_receiver_(
|
| - RtpReceiver::CreateAudioReceiver(VoEModuleId(instanceId, channelId),
|
| - Clock::GetRealTimeClock(),
|
| + RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
|
| this,
|
| this,
|
| this,
|
| @@ -817,7 +793,6 @@ Channel::Channel(int32_t channelId,
|
| _outputAudioLevel.Clear();
|
|
|
| RtpRtcp::Configuration configuration;
|
| - configuration.id = VoEModuleId(instanceId, channelId);
|
| configuration.audio = true;
|
| configuration.outgoing_transport = this;
|
| configuration.audio_messages = this;
|
| @@ -3890,11 +3865,11 @@ Channel::SendPacketRaw(const void *data, size_t len, bool RTCP)
|
| }
|
| if (!RTCP)
|
| {
|
| - return _transportPtr->SendPacket(_channelId, data, len);
|
| + return _transportPtr->SendPacket(data, len);
|
| }
|
| else
|
| {
|
| - return _transportPtr->SendRTCPPacket(_channelId, data, len);
|
| + return _transportPtr->SendRTCPPacket(data, len);
|
| }
|
| }
|
|
|
|
|