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Bug 864654: cleanup AudioConduit r=ekr

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This commit is contained in:
Randell Jesup 2013-10-23 06:20:55 -04:00
parent 50c7d4e360
commit 3ad7a7e6d1
2 changed files with 50 additions and 51 deletions
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66
media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
View File

@ -42,7 +42,7 @@ mozilla::RefPtr<AudioSessionConduit> AudioSessionConduit::Create(AudioSessionCon
{
CSFLogError(logTag, "%s AudioConduit Init Failed ", __FUNCTION__);
delete obj;
return NULL;
return nullptr;
}
CSFLogDebug(logTag, "%s Successfully created AudioConduit ", __FUNCTION__);
return obj;
@ -110,7 +110,7 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
if (mOtherDirection)
{
// mOtherDirection owns these now!
mOtherDirection->mOtherDirection = NULL;
mOtherDirection->mOtherDirection = nullptr;
// let other side we terminated the channel
mOtherDirection->mShutDown = true;
mVoiceEngine = nullptr;
@ -283,7 +283,7 @@ WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
if(mPtrVoEBase->StopSend(mChannel) == -1)
{
CSFLogError(logTag, "%s StopSend() Failed %d ", __FUNCTION__,
mPtrVoEBase->LastError());
mPtrVoEBase->LastError());
return kMediaConduitUnknownError;
}
}
@ -304,9 +304,11 @@ WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
if(error == VE_CANNOT_SET_SEND_CODEC || error == VE_CODEC_ERROR)
{
CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);
return kMediaConduitInvalidSendCodec;
}
CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,
mPtrVoEBase->LastError());
return kMediaConduitUnknownError;
}
@ -357,7 +359,6 @@ WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
codecConfig->mChannels,
codecConfig->mRate);
mEngineTransmitting = true;
return kMediaConduitNoError;
}
@ -371,13 +372,13 @@ WebrtcAudioConduit::ConfigureRecvMediaCodecs(
int error = 0; //webrtc engine errors
bool success = false;
// are we receiving already. If so, stop receiving and playout
// since we can't apply new recv codec when the engine is playing
// Are we receiving already? If so, stop receiving and playout
// since we can't apply new recv codec when the engine is playing.
if(mEngineReceiving)
{
CSFLogDebug(logTag, "%s Engine Already Receiving. Attemping to Stop ", __FUNCTION__);
// AudioEngine doesn't fail fatal on stop reception. Ref:voe_errors.h.
// hence we need-not be strict in failing here on error
// AudioEngine doesn't fail fatally on stopping reception. Ref:voe_errors.h.
// hence we need not be strict in failing here on errors
mPtrVoEBase->StopReceive(mChannel);
CSFLogDebug(logTag, "%s Attemping to Stop playout ", __FUNCTION__);
if(mPtrVoEBase->StopPlayout(mChannel) == -1)
@ -392,13 +393,15 @@ WebrtcAudioConduit::ConfigureRecvMediaCodecs(
mEngineReceiving = false;
if(!codecConfigList.size())
if(codecConfigList.empty())
{
CSFLogError(logTag, "%s Zero number of codecs to configure", __FUNCTION__);
return kMediaConduitMalformedArgument;
}
//Try Applying the codecs in the list
// Try Applying the codecs in the list.
// We succeed if at least one codec was applied and reception was
// started successfully.
for(std::vector<AudioCodecConfig*>::size_type i=0 ;i<codecConfigList.size();i++)
{
//if the codec param is invalid or diplicate, return error
@ -435,7 +438,6 @@ WebrtcAudioConduit::ConfigureRecvMediaCodecs(
} //end for
//Success == false indicates none of the codec was applied
if(!success)
{
CSFLogError(logTag, "%s Setting Receive Codec Failed ", __FUNCTION__);
@ -488,7 +490,7 @@ WebrtcAudioConduit::SendAudioFrame(const int16_t audio_data[],
(IsSamplingFreqSupported(samplingFreqHz) == false) ||
((lengthSamples % (samplingFreqHz / 100) != 0)) )
{
CSFLogError(logTag, "%s Invalid Params ", __FUNCTION__);
CSFLogError(logTag, "%s Invalid Parameters ",__FUNCTION__);
MOZ_ASSERT(PR_FALSE);
return kMediaConduitMalformedArgument;
}
@ -610,11 +612,10 @@ WebrtcAudioConduit::ReceivedRTPPacket(const void *data, int len)
return kMediaConduitUnknownError;
}
} else {
//engine not receiving
CSFLogError(logTag, "%s ReceivedRTPPacket: Engine Error", __FUNCTION__);
CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
//good here
return kMediaConduitNoError;
}
@ -636,19 +637,16 @@ WebrtcAudioConduit::ReceivedRTCPPacket(const void *data, int len)
return kMediaConduitUnknownError;
}
} else {
//engine not running
CSFLogError(logTag, "%s ReceivedRTPPacket: Engine Error", __FUNCTION__);
CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
//good here
return kMediaConduitNoError;
}
//WebRTC::RTP Callback Implementation
int WebrtcAudioConduit::SendPacket(int channel, const void* data, int len)
{
CSFLogDebug(logTag, "%s : channel %d %s",__FUNCTION__,channel,
CSFLogDebug(logTag, "%s : channel %d %s", __FUNCTION__, channel,
(mEngineReceiving && mOtherDirection) ? "(using mOtherDirection)" : "");
if (mEngineReceiving)
@ -682,18 +680,18 @@ int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, int len)
{
return mOtherDirection->SendRTCPPacket(channel, data, len);
}
CSFLogDebug(logTag, "%s : Asked to send RTCP without an RTP receiver on channel %d",
__FUNCTION__, channel);
return -1;
}
// We come here if we have only one pipeline/conduit setup,
// such as for unidirectional streams.
// We also end up here if we are receiving
if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
{
CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
return len;
} else {
if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
{
CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
return len;
} else {
CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
return -1;
}
CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
return -1;
}
}
@ -812,12 +810,12 @@ WebrtcAudioConduit::CheckCodecForMatch(const AudioCodecConfig* codecInfo) const
/**
* Perform validation on the codecCofig to be applied
* Perform validation on the codecConfig to be applied.
* Verifies if the codec is already applied.
*/
MediaConduitErrorCode
WebrtcAudioConduit::ValidateCodecConfig(const AudioCodecConfig* codecInfo,
bool send) const
bool send) const
{
bool codecAppliedAlready = false;

35
media/webrtc/signaling/src/media-conduit/AudioConduit.h
View File

@ -48,13 +48,13 @@ public:
* APIs used by the registered external transport to this Conduit to
* feed in received RTP Frames to the VoiceEngine for decoding
*/
virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
/**
* APIs used by the registered external transport to this Conduit to
* feed in received RTCP Frames to the VoiceEngine for decoding
*/
virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
/**
* Function to configure send codec for the audio session
@ -64,8 +64,7 @@ public:
* NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
* transmission sub-system on the engine.
*/
virtual MediaConduitErrorCode ConfigureSendMediaCodec(
const AudioCodecConfig* codecConfig);
virtual MediaConduitErrorCode ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig);
/**
* Function to configure list of receive codecs for the audio session
* @param sendSessionConfig: CodecConfiguration
@ -75,15 +74,14 @@ public:
* NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
* transmission sub-system on the engine.
*/
virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
const std::vector<AudioCodecConfig* >& codecConfigList);
virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
const std::vector<AudioCodecConfig* >& codecConfigList);
/**
* Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEnigne
* Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEngine
* shall be passed to the registered transport for transporting externally.
*/
virtual MediaConduitErrorCode AttachTransport(
mozilla::RefPtr<TransportInterface> aTransport);
virtual MediaConduitErrorCode AttachTransport(mozilla::RefPtr<TransportInterface> aTransport);
/**
* Function to deliver externally captured audio sample for encoding and transport
@ -101,10 +99,10 @@ public:
* This ensures the inserted audio-samples can be transmitted by the conduit
*
*/
virtual MediaConduitErrorCode SendAudioFrame(const int16_t speechData[],
int32_t lengthSamples,
int32_t samplingFreqHz,
int32_t capture_time);
virtual MediaConduitErrorCode SendAudioFrame(const int16_t speechData[],
int32_t lengthSamples,
int32_t samplingFreqHz,
int32_t capture_time);
/**
* Function to grab a decoded audio-sample from the media engine for rendering
@ -143,14 +141,14 @@ public:
WebrtcAudioConduit():
mOtherDirection(NULL),
mOtherDirection(nullptr),
mShutDown(false),
mVoiceEngine(NULL),
mTransport(NULL),
mVoiceEngine(nullptr),
mTransport(nullptr),
mEngineTransmitting(false),
mEngineReceiving(false),
mChannel(-1),
mCurSendCodecConfig(NULL),
mCurSendCodecConfig(nullptr),
mCaptureDelay(150),
mEchoOn(true),
mEchoCancel(webrtc::kEcAec)
@ -195,6 +193,9 @@ private:
//Utility function to dump recv codec database
void DumpCodecDB() const;
// The two sides of a send/receive pair of conduits each keep a pointer to the other.
// The also share a single VoiceEngine and mChannel. Shutdown must be coordinated
// carefully to avoid double-freeing or accessing after one frees.
WebrtcAudioConduit* mOtherDirection;
// Other side has shut down our channel and related items already
bool mShutDown;