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