| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 95b08e71587e2aa970834b9c60b9fe41d9907d66..b78d73ad0b0ffa10f4f03a1b6a3bd6813dce02ad 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -520,14 +520,18 @@ bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
|
| return WebRtcVoiceCodecs::ToCodecInst(in, out);
|
| }
|
|
|
| -WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm)
|
| - : WebRtcVoiceEngine(adm, new VoEWrapper()) {
|
| +WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| + webrtc::AudioDeviceModule* adm,
|
| + const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
|
| + : WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) {
|
| audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe()));
|
| }
|
|
|
| -WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
|
| - VoEWrapper* voe_wrapper)
|
| - : adm_(adm), voe_wrapper_(voe_wrapper) {
|
| +WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| + webrtc::AudioDeviceModule* adm,
|
| + const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
|
| + VoEWrapper* voe_wrapper)
|
| + : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
| RTC_DCHECK(voe_wrapper);
|
| @@ -547,7 +551,8 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
|
| webrtc::Trace::SetTraceCallback(this);
|
| webrtc::Trace::set_level_filter(kElevatedTraceFilter);
|
| LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
|
| - RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get()));
|
| + RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
|
| + decoder_factory_));
|
| webrtc::Trace::set_level_filter(kDefaultTraceFilter);
|
|
|
| // No ADM supplied? Get the default one from VoE.
|
| @@ -1275,14 +1280,16 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
|
|
|
| class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
|
| public:
|
| - WebRtcAudioReceiveStream(int ch,
|
| - uint32_t remote_ssrc,
|
| - uint32_t local_ssrc,
|
| - bool use_transport_cc,
|
| - const std::string& sync_group,
|
| - const std::vector<webrtc::RtpExtension>& extensions,
|
| - webrtc::Call* call,
|
| - webrtc::Transport* rtcp_send_transport)
|
| + WebRtcAudioReceiveStream(
|
| + int ch,
|
| + uint32_t remote_ssrc,
|
| + uint32_t local_ssrc,
|
| + bool use_transport_cc,
|
| + const std::string& sync_group,
|
| + const std::vector<webrtc::RtpExtension>& extensions,
|
| + webrtc::Call* call,
|
| + webrtc::Transport* rtcp_send_transport,
|
| + const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
|
| : call_(call), config_() {
|
| RTC_DCHECK_GE(ch, 0);
|
| RTC_DCHECK(call);
|
| @@ -1291,6 +1298,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
|
| config_.rtcp_send_transport = rtcp_send_transport;
|
| config_.voe_channel_id = ch;
|
| config_.sync_group = sync_group;
|
| + config_.decoder_factory = decoder_factory;
|
| RecreateAudioReceiveStream(use_transport_cc, extensions);
|
| }
|
|
|
| @@ -2168,7 +2176,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
| ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
|
| recv_transport_cc_enabled_,
|
| sp.sync_label, recv_rtp_extensions_,
|
| - call_, this)));
|
| + call_, this,
|
| + engine()->decoder_factory_)));
|
|
|
| SetNack(channel, send_codec_spec_.nack_enabled);
|
| SetPlayout(channel, playout_);
|
|
|