Log Call {audio, video} stream deletions.
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1400333002
Cr-Commit-Position: refs/heads/master@{#10286}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 064749c..c725e37 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -13,6 +13,7 @@
#include <string>
#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 @@
: 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() {
+ 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 70ad4fc..1e52724 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -24,7 +24,7 @@
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 @@
// 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 41b2c83..0ccfb61 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -176,16 +176,19 @@
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 @@
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 @@
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 0d0953e..141e918 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -139,6 +139,7 @@
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() {
+ 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 d55adf0..af6ae8e 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -117,6 +117,7 @@
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() {
+ LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
vie_channel_->RegisterSendFrameCountObserver(nullptr);
vie_channel_->RegisterSendBitrateObserver(nullptr);
vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
