| Index: webrtc/voice_engine/channel.cc
|
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
|
| index fa44785344b222ed834710edce4df83151a6005a..ee170587325b479a2e460327d7dfd11b597c70fa 100644
|
| --- a/webrtc/voice_engine/channel.cc
|
| +++ b/webrtc/voice_engine/channel.cc
|
| @@ -214,8 +214,8 @@ Channel::OnRxVadDetected(int vadDecision)
|
| return 0;
|
| }
|
|
|
| -int
|
| -Channel::SendPacket(const void *data, size_t len)
|
| +bool
|
| +Channel::SendRtp(const uint8_t *data, size_t len)
|
| {
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
| "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
|
| @@ -227,55 +227,54 @@ Channel::SendPacket(const void *data, size_t len)
|
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
|
| "Channel::SendPacket() failed to send RTP packet due to"
|
| " invalid transport object");
|
| - return -1;
|
| + return false;
|
| }
|
|
|
| uint8_t* bufferToSendPtr = (uint8_t*)data;
|
| size_t bufferLength = len;
|
|
|
| - int n = _transportPtr->SendPacket(bufferToSendPtr, bufferLength);
|
| - if (n < 0) {
|
| + if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength)) {
|
| std::string transport_name =
|
| _externalTransport ? "external transport" : "WebRtc sockets";
|
| WEBRTC_TRACE(kTraceError, kTraceVoice,
|
| VoEId(_instanceId,_channelId),
|
| "Channel::SendPacket() RTP transmission using %s failed",
|
| transport_name.c_str());
|
| - return -1;
|
| + return false;
|
| }
|
| - return n;
|
| + return true;
|
| }
|
|
|
| -int
|
| -Channel::SendRTCPPacket(const void *data, size_t len)
|
| +bool
|
| +Channel::SendRtcp(const uint8_t *data, size_t len)
|
| {
|
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
|
| - "Channel::SendRTCPPacket(len=%" PRIuS ")", len);
|
| + "Channel::SendRtcp(len=%" PRIuS ")", len);
|
|
|
| CriticalSectionScoped cs(&_callbackCritSect);
|
| if (_transportPtr == NULL)
|
| {
|
| WEBRTC_TRACE(kTraceError, kTraceVoice,
|
| VoEId(_instanceId,_channelId),
|
| - "Channel::SendRTCPPacket() failed to send RTCP packet"
|
| + "Channel::SendRtcp() failed to send RTCP packet"
|
| " due to invalid transport object");
|
| - return -1;
|
| + return false;
|
| }
|
|
|
| uint8_t* bufferToSendPtr = (uint8_t*)data;
|
| size_t bufferLength = len;
|
|
|
| - int n = _transportPtr->SendRTCPPacket(bufferToSendPtr, bufferLength);
|
| + int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
|
| if (n < 0) {
|
| std::string transport_name =
|
| _externalTransport ? "external transport" : "WebRtc sockets";
|
| WEBRTC_TRACE(kTraceInfo, kTraceVoice,
|
| VoEId(_instanceId,_channelId),
|
| - "Channel::SendRTCPPacket() transmission using %s failed",
|
| + "Channel::SendRtcp() transmission using %s failed",
|
| transport_name.c_str());
|
| - return -1;
|
| + return false;
|
| }
|
| - return n;
|
| + return true;
|
| }
|
|
|
| void Channel::OnPlayTelephoneEvent(uint8_t event,
|
| @@ -3708,24 +3707,6 @@ Channel::InsertInbandDtmfTone()
|
| return 0;
|
| }
|
|
|
| -int32_t
|
| -Channel::SendPacketRaw(const void *data, size_t len, bool RTCP)
|
| -{
|
| - CriticalSectionScoped cs(&_callbackCritSect);
|
| - if (_transportPtr == NULL)
|
| - {
|
| - return -1;
|
| - }
|
| - if (!RTCP)
|
| - {
|
| - return _transportPtr->SendPacket(data, len);
|
| - }
|
| - else
|
| - {
|
| - return _transportPtr->SendRTCPPacket(data, len);
|
| - }
|
| -}
|
| -
|
| void Channel::UpdatePlayoutTimestamp(bool rtcp) {
|
| uint32_t playout_timestamp = 0;
|
|
|
|
|