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; |