diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 064749c61e..c725e37477 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -13,6 +13,7 @@ #include #include "webrtc/base/checks.h" +#include "webrtc/base/logging.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/system_wrappers/interface/tick_util.h" @@ -48,6 +49,7 @@ AudioReceiveStream::AudioReceiveStream( : remote_bitrate_estimator_(remote_bitrate_estimator), config_(config), rtp_header_parser_(RtpHeaderParser::Create()) { + LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); RTC_DCHECK(config.voe_channel_id != -1); RTC_DCHECK(remote_bitrate_estimator_ != nullptr); RTC_DCHECK(rtp_header_parser_ != nullptr); @@ -70,10 +72,18 @@ AudioReceiveStream::AudioReceiveStream( } } +AudioReceiveStream::~AudioReceiveStream() { + LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); +} + webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { return webrtc::AudioReceiveStream::Stats(); } +const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { + return config_; +} + void AudioReceiveStream::Start() { } diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index 70ad4fcf2b..1e52724020 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -24,7 +24,7 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream { public: AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, const webrtc::AudioReceiveStream::Config& config); - ~AudioReceiveStream() override {} + ~AudioReceiveStream() override; // webrtc::ReceiveStream implementation. void Start() override; @@ -38,9 +38,7 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream { // webrtc::AudioReceiveStream implementation. webrtc::AudioReceiveStream::Stats GetStats() const override; - const webrtc::AudioReceiveStream::Config& config() const { - return config_; - } + const webrtc::AudioReceiveStream::Config& config() const; private: RemoteBitrateEstimator* const remote_bitrate_estimator_; diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 41b2c83d38..0ccfb611e4 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -176,16 +176,19 @@ PacketReceiver* Call::Receiver() { return this; } webrtc::AudioSendStream* Call::CreateAudioSendStream( const webrtc::AudioSendStream::Config& config) { + // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config + // logging to AudioSendStream constructor. return nullptr; } void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { + // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config + // logging to AudioSendStream destructor. } webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( const webrtc::AudioReceiveStream::Config& config) { TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); - LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString(); AudioReceiveStream* receive_stream = new AudioReceiveStream( channel_group_->GetRemoteBitrateEstimator(), config); { @@ -224,8 +227,6 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream( const webrtc::VideoSendStream::Config& config, const VideoEncoderConfig& encoder_config) { TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); - LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString(); - RTC_DCHECK(!config.rtp.ssrcs.empty()); // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if // the call has already started. @@ -288,7 +289,6 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( const webrtc::VideoReceiveStream::Config& config) { TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); - LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString(); VideoReceiveStream* receive_stream = new VideoReceiveStream( num_cpu_cores_, channel_group_.get(), rtc::AtomicOps::Increment(&next_channel_id_), config, diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc index 0d0953ea91..141e918b7f 100644 --- a/webrtc/video/video_receive_stream.cc +++ b/webrtc/video/video_receive_stream.cc @@ -139,6 +139,7 @@ VideoReceiveStream::VideoReceiveStream(int num_cpu_cores, clock_(Clock::GetRealTimeClock()), channel_group_(channel_group), channel_id_(channel_id) { + LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); RTC_CHECK(channel_group_->CreateReceiveChannel( channel_id_, &transport_adapter_, num_cpu_cores, config)); @@ -257,6 +258,7 @@ VideoReceiveStream::VideoReceiveStream(int num_cpu_cores, } VideoReceiveStream::~VideoReceiveStream() { + LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString(); incoming_video_stream_->Stop(); vie_channel_->RegisterPreRenderCallback(nullptr); vie_channel_->RegisterPreDecodeImageCallback(nullptr); diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc index d55adf0125..af6ae8e4aa 100644 --- a/webrtc/video/video_send_stream.cc +++ b/webrtc/video/video_send_stream.cc @@ -117,6 +117,7 @@ VideoSendStream::VideoSendStream( channel_id_(channel_id), use_config_bitrate_(true), stats_proxy_(Clock::GetRealTimeClock(), config) { + LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); RTC_DCHECK(!config_.rtp.ssrcs.empty()); RTC_CHECK(channel_group->CreateSendChannel( channel_id_, &transport_adapter_, &stats_proxy_, @@ -194,6 +195,7 @@ VideoSendStream::VideoSendStream( } VideoSendStream::~VideoSendStream() { + LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString(); vie_channel_->RegisterSendFrameCountObserver(nullptr); vie_channel_->RegisterSendBitrateObserver(nullptr); vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);