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); |
} |
} |