Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
BUG=webrtc:6195
Review-Url: https://codereview.webrtc.org/2226823003
Cr-Commit-Position: refs/heads/master@{#14331}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 417720c..ee84b96 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -63,7 +63,8 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
CongestionController* congestion_controller,
- BitrateAllocator* bitrate_allocator)
+ BitrateAllocator* bitrate_allocator,
+ RtcEventLog* event_log)
: worker_queue_(worker_queue),
config_(config),
audio_state_(audio_state),
@@ -75,6 +76,7 @@
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+ channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->RegisterSenderCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(),
@@ -107,6 +109,7 @@
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
+ channel_proxy_->SetRtcEventLog(nullptr);
}
void AudioSendStream::Start() {
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index e92c326..2e0b7ae 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -22,6 +22,7 @@
namespace webrtc {
class CongestionController;
class VoiceEngine;
+class RtcEventLog;
namespace voe {
class ChannelProxy;
@@ -35,7 +36,8 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
CongestionController* congestion_controller,
- BitrateAllocator* bitrate_allocator);
+ BitrateAllocator* bitrate_allocator,
+ RtcEventLog* event_log);
~AudioSendStream() override;
// webrtc::AudioSendStream implementation.
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 9172064..ebac828 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -106,6 +106,10 @@
.Times(1);
EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
.Times(1);
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
+ .Times(1); // Destructor resets the event log
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
@@ -128,6 +132,7 @@
}
BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
rtc::TaskQueue* worker_queue() { return &worker_queue_; }
+ RtcEventLog* event_log() { return &event_log_; }
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
@@ -210,14 +215,16 @@
ConfigHelper helper;
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.congestion_controller(), helper.bitrate_allocator());
+ helper.congestion_controller(), helper.bitrate_allocator(),
+ helper.event_log());
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper;
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.congestion_controller(), helper.bitrate_allocator());
+ helper.congestion_controller(), helper.bitrate_allocator(),
+ helper.event_log());
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventCode, kTelephoneEventDuration));
@@ -227,7 +234,8 @@
ConfigHelper helper;
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.congestion_controller(), helper.bitrate_allocator());
+ helper.congestion_controller(), helper.bitrate_allocator(),
+ helper.event_log());
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
send_stream.SetMuted(true);
}
@@ -236,7 +244,8 @@
ConfigHelper helper;
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.congestion_controller(), helper.bitrate_allocator());
+ helper.congestion_controller(), helper.bitrate_allocator(),
+ helper.event_log());
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -265,7 +274,8 @@
ConfigHelper helper;
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
- helper.congestion_controller(), helper.bitrate_allocator());
+ helper.congestion_controller(), helper.bitrate_allocator(),
+ helper.event_log());
helper.SetupMockForGetStats();
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index a3b4ec6..16a6f46 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -380,7 +380,7 @@
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
- bitrate_allocator_.get());
+ bitrate_allocator_.get(), event_log_.get());
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
