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Unified Diff: webrtc/voice_engine/channel.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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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);
}
}
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