This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/
- Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
TBR=mflodman@webrtc.org
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 17979d5..417720c 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -16,7 +16,9 @@
#include "webrtc/audio/conversion.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/task_queue.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -59,9 +61,11 @@
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ rtc::TaskQueue* worker_queue,
CongestionController* congestion_controller,
BitrateAllocator* bitrate_allocator)
- : config_(config),
+ : worker_queue_(worker_queue),
+ config_(config),
audio_state_(audio_state),
bitrate_allocator_(bitrate_allocator) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
@@ -109,8 +113,13 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
- bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
- config_.max_bitrate_kbps * 1000, 0, true);
+ rtc::Event thread_sync_event(false /* manual_reset */, false);
+ worker_queue_->PostTask([this, &thread_sync_event] {
+ bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
+ config_.max_bitrate_kbps * 1000, 0, true);
+ thread_sync_event.Set();
+ });
+ thread_sync_event.Wait(rtc::Event::kForever);
}
ScopedVoEInterface<VoEBase> base(voice_engine());
@@ -122,7 +131,13 @@
void AudioSendStream::Stop() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- bitrate_allocator_->RemoveObserver(this);
+ rtc::Event thread_sync_event(false /* manual_reset */, false);
+ worker_queue_->PostTask([this, &thread_sync_event] {
+ bitrate_allocator_->RemoveObserver(this);
+ thread_sync_event.Set();
+ });
+ thread_sync_event.Wait(rtc::Event::kForever);
+
ScopedVoEInterface<VoEBase> base(voice_engine());
int error = base->StopSend(config_.voe_channel_id);
if (error != 0) {
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index ec2a4db..e92c326 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -33,6 +33,7 @@
public:
AudioSendStream(const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ rtc::TaskQueue* worker_queue,
CongestionController* congestion_controller,
BitrateAllocator* bitrate_allocator);
~AudioSendStream() override;
@@ -59,6 +60,7 @@
VoiceEngine* voice_engine() const;
rtc::ThreadChecker thread_checker_;
+ rtc::TaskQueue* worker_queue_;
const webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 7f940fc..9172064 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -16,6 +16,7 @@
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/base/task_queue.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
#include "webrtc/call/mock/mock_rtc_event_log.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
@@ -65,7 +66,8 @@
&bitrate_observer_,
&remote_bitrate_observer_,
&event_log_),
- bitrate_allocator_(&limit_observer_) {
+ bitrate_allocator_(&limit_observer_),
+ worker_queue_("ConfigHelper_worker_queue") {
using testing::Invoke;
using testing::StrEq;
@@ -125,6 +127,7 @@
return &congestion_controller_;
}
BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
+ rtc::TaskQueue* worker_queue() { return &worker_queue_; }
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
@@ -181,6 +184,9 @@
MockRtcEventLog event_log_;
testing::NiceMock<MockLimitObserver> limit_observer_;
BitrateAllocator bitrate_allocator_;
+ // |worker_queue| is defined last to ensure all pending tasks are cancelled
+ // and deleted before any other members.
+ rtc::TaskQueue worker_queue_;
};
} // namespace
@@ -202,16 +208,16 @@
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
- helper.congestion_controller(),
- helper.bitrate_allocator());
+ internal::AudioSendStream send_stream(
+ helper.config(), helper.audio_state(), helper.worker_queue(),
+ helper.congestion_controller(), helper.bitrate_allocator());
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
- helper.congestion_controller(),
- helper.bitrate_allocator());
+ internal::AudioSendStream send_stream(
+ helper.config(), helper.audio_state(), helper.worker_queue(),
+ helper.congestion_controller(), helper.bitrate_allocator());
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventCode, kTelephoneEventDuration));
@@ -219,18 +225,18 @@
TEST(AudioSendStreamTest, SetMuted) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
- helper.congestion_controller(),
- helper.bitrate_allocator());
+ internal::AudioSendStream send_stream(
+ helper.config(), helper.audio_state(), helper.worker_queue(),
+ helper.congestion_controller(), helper.bitrate_allocator());
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
send_stream.SetMuted(true);
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
- helper.congestion_controller(),
- helper.bitrate_allocator());
+ internal::AudioSendStream send_stream(
+ helper.config(), helper.audio_state(), helper.worker_queue(),
+ helper.congestion_controller(), helper.bitrate_allocator());
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -257,9 +263,9 @@
TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
ConfigHelper helper;
- internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
- helper.congestion_controller(),
- helper.bitrate_allocator());
+ internal::AudioSendStream send_stream(
+ helper.config(), helper.audio_state(), helper.worker_queue(),
+ helper.congestion_controller(), helper.bitrate_allocator());
helper.SetupMockForGetStats();
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);