commit 26091b1118e37b5cbbfe0de9869127fd5c1b01d0
author perkj <perkj@webrtc.org> Thu Sep 01 01:17:40 2016 -0700
committer Commit bot <commit-bot@chromium.org> Thu Sep 01 08:17:43 2016 +0000
tree dc50c0b97bbbc9035467f8842ba4492c47a6bc8b
parent 9f8e37b6f833cb6c821c6f401421b7c33cbe4b27

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}

webrtc/audio/audio_send_stream.cc

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) {