bool WebrtcAudioConduit::SetLocalSSRC(unsigned int ssrc)
{
unsigned int oldSsrc;
if (!GetLocalSSRC(&oldSsrc)) {
MOZ_ASSERT(false, "GetLocalSSRC failed");
return false;
}
if (oldSsrc == ssrc) {
return true;
}
bool wasTransmitting = mEngineTransmitting;
if (StopTransmitting() != kMediaConduitNoError) {
return false;
}
if (mPtrRTP->SetLocalSSRC(mChannel, ssrc)) {
return false;
}
if (wasTransmitting) {
if (StartTransmitting() != kMediaConduitNoError) {
return false;
}
}
return true;
}
/**
* Note: Setting the send-codec on the Video Engine will restart the encoder,
* sets up new SSRC and reset RTP_RTCP module with the new codec setting.
*
* Note: this is called from MainThread, and the codec settings are read on
* videoframe delivery threads (i.e in SendVideoFrame(). With
* renegotiation/reconfiguration, this now needs a lock! Alternatively
* changes could be queued until the next frame is delivered using an
* Atomic pointer and swaps.
*/
MediaConduitErrorCode
WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
{
CSFLogDebug(logTag, "%s for %s", __FUNCTION__, codecConfig ? codecConfig->mName.c_str() : "<null>");
bool codecFound = false;
MediaConduitErrorCode condError = kMediaConduitNoError;
int error = 0; //webrtc engine errors
webrtc::VideoCodec video_codec;
std::string payloadName;
memset(&video_codec, 0, sizeof(video_codec));
{
//validate basic params
if((condError = ValidateCodecConfig(codecConfig,true)) != kMediaConduitNoError)
{
return condError;
}
}
condError = StopTransmitting();
if (condError != kMediaConduitNoError) {
return condError;
}
if (mExternalSendCodec &&
codecConfig->mType == mExternalSendCodec->mType) {
CSFLogError(logTag, "%s Configuring External H264 Send Codec", __FUNCTION__);
// width/height will be overridden on the first frame
video_codec.width = 320;
video_codec.height = 240;
#ifdef MOZ_WEBRTC_OMX
if (codecConfig->mType == webrtc::kVideoCodecH264) {
video_codec.resolution_divisor = 16;
} else {
video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions
}
#else
video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions
#endif
video_codec.qpMax = 56;
video_codec.numberOfSimulcastStreams = 1;
video_codec.mode = webrtc::kRealtimeVideo;
codecFound = true;
} else {
// we should be good here to set the new codec.
for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
{
if(0 == mPtrViECodec->GetCodec(idx, video_codec))
{
payloadName = video_codec.plName;
if(codecConfig->mName.compare(payloadName) == 0)
{
// Note: side-effect of this is that video_codec is filled in
// by GetCodec()
codecFound = true;
break;
}
}
}//for
}
if(codecFound == false)
{
CSFLogError(logTag, "%s Codec Mismatch ", __FUNCTION__);
return kMediaConduitInvalidSendCodec;
}
// Note: only for overriding parameters from GetCodec()!
CodecConfigToWebRTCCodec(codecConfig, video_codec);
if(mPtrViECodec->SetSendCodec(mChannel, video_codec) == -1)
{
error = mPtrViEBase->LastError();
if(error == kViECodecInvalidCodec)
{
CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);
return kMediaConduitInvalidSendCodec;
}
CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitUnknownError;
}
if (!mVideoCodecStat) {
mVideoCodecStat = new VideoCodecStatistics(mChannel, mPtrViECodec);
}
mVideoCodecStat->Register(true);
mSendingWidth = 0;
mSendingHeight = 0;
mSendingFramerate = video_codec.maxFramerate;
if(codecConfig->RtcpFbNackIsSet("")) {
CSFLogDebug(logTag, "Enabling NACK (send) for video stream\n");
if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
{
CSFLogError(logTag, "%s NACKStatus Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitNACKStatusError;
}
}
condError = StartTransmitting();
if (condError != kMediaConduitNoError) {
return condError;
}
{
MutexAutoLock lock(mCodecMutex);
//Copy the applied config for future reference.
mCurSendCodecConfig = new VideoCodecConfig(*codecConfig);
}
mPtrRTP->SetRembStatus(mChannel, true, false);
return kMediaConduitNoError;
}
MediaConduitErrorCode
WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
{
CSFLogDebug(logTag, "%s ", __FUNCTION__);
MediaConduitErrorCode condError = kMediaConduitNoError;
int error = 0;//webrtc engine errors
webrtc::CodecInst cinst;
//validate codec param
if((condError = ValidateCodecConfig(codecConfig, true)) != kMediaConduitNoError)
{
return condError;
}
condError = StopTransmitting();
if (condError != kMediaConduitNoError) {
return condError;
}
if(!CodecConfigToWebRTCCodec(codecConfig,cinst))
{
CSFLogError(logTag,"%s CodecConfig to WebRTC Codec Failed ",__FUNCTION__);
return kMediaConduitMalformedArgument;
}
if(mPtrVoECodec->SetSendCodec(mChannel, cinst) == -1)
{
error = mPtrVoEBase->LastError();
CSFLogError(logTag, "%s SetSendCodec - Invalid Codec %d ",__FUNCTION__,
error);
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;
}
#if !defined(MOZILLA_EXTERNAL_LINKAGE)
// TEMPORARY - see bug 694814 comment 2
nsresult rv;
nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
if (NS_SUCCEEDED(rv)) {
nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
if (branch) {
branch->GetIntPref("media.peerconnection.capture_delay", &mCaptureDelay);
}
}
#endif
condError = StartTransmitting();
if (condError != kMediaConduitNoError) {
return condError;
}
//Copy the applied config for future reference.
delete mCurSendCodecConfig;
mCurSendCodecConfig = new AudioCodecConfig(codecConfig->mType,
codecConfig->mName,
codecConfig->mFreq,
codecConfig->mPacSize,
codecConfig->mChannels,
codecConfig->mRate);
return kMediaConduitNoError;
}
