Onion soup content/renderer/media/webrtc/webrtc_audio_sink.cc|h

This is part of phase 1.9 on the design document [1].

[1] https://docs.google.com/document/d/1AJKVA5U4nDkyDB9p4ROrggWXadCxyy-grKaE9KS5vOU/

Since there are clients in //content directly referencing to this class,
it was temporarily placed in blink/public/platform.

Also, once its clients get moved to Blink, and so this API, it can
switch over from STL types (std::string and vector) to its WTF
counterparts. TODO is left.

BUG=787254
R=guidou@chromium.org, haraken@chromium.org

Change-Id: I3519318d5ae6f5b3641442f78d5e6116801dbbe9
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1778005
Commit-Queue: Antonio Gomes <tonikitoo@igalia.com>
Reviewed-by: Kentaro Hara <haraken@chromium.org>
Cr-Commit-Position: refs/heads/master@{#692180}

content/renderer/BUILD.gn

@@ -230,8 +230,6 @@ target(link_target_type, "renderer") {
     "media/webrtc/transmission_encoding_info_handler.h",
     "media/webrtc/video_codec_factory.cc",
     "media/webrtc/video_codec_factory.h",
-    "media/webrtc/webrtc_audio_sink.cc",
-    "media/webrtc/webrtc_audio_sink.h",
     "media/webrtc/webrtc_media_stream_track_adapter.cc",
     "media/webrtc/webrtc_media_stream_track_adapter.h",
     "media/webrtc/webrtc_media_stream_track_adapter_map.cc",
@@ -510,7 +508,6 @@ target(link_target_type, "renderer") {
     "//third_party/webrtc/p2p:libstunprober",
     "//third_party/webrtc/p2p:rtc_p2p",
     "//third_party/webrtc/pc:libjingle_peerconnection",
-    "//third_party/webrtc/pc:peerconnection",
     "//third_party/webrtc/pc:rtc_pc",
     "//third_party/webrtc/pc:rtc_pc_base",
     "//third_party/webrtc/rtc_base:async_resolver_interface",

content/renderer/media/webrtc/webrtc_media_stream_track_adapter.cc

@@ -164,9 +164,9 @@ void WebRtcMediaStreamTrackAdapter::InitializeLocalAudioTrack(
   // the webrtc::AudioSourceInterface, and also do not need references to the
   // audio level calculator or audio processor passed to the sink.
   webrtc::AudioSourceInterface* source_interface = nullptr;
-  local_track_audio_sink_.reset(
-      new WebRtcAudioSink(web_track_.Id().Utf8(), source_interface,
-                          factory_->GetWebRtcSignalingThread(), main_thread_));
+  local_track_audio_sink_.reset(new blink::WebRtcAudioSink(
+      web_track_.Id().Utf8(), source_interface,
+      factory_->GetWebRtcSignalingThread(), main_thread_));
 
   if (auto* media_stream_source = blink::ProcessedLocalAudioSource::From(
           blink::MediaStreamAudioSource::From(web_track_.Source()))) {

content/renderer/media/webrtc/webrtc_media_stream_track_adapter.h

@@ -11,8 +11,8 @@
 #include "base/synchronization/waitable_event.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h"
-#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_track_adapter.h"
+#include "third_party/blink/public/platform/modules/peerconnection/webrtc_audio_sink.h"
 #include "third_party/blink/public/platform/web_media_stream.h"
 #include "third_party/blink/public/platform/web_media_stream_track.h"
 #include "third_party/blink/public/web/modules/mediastream/remote_media_stream_track_adapter.h"
@@ -64,7 +64,7 @@ class CONTENT_EXPORT WebRtcMediaStreamTrackAdapter
   bool IsEqual(const blink::WebMediaStreamTrack& web_track);
 
   // For testing.
-  WebRtcAudioSink* GetLocalTrackAudioSinkForTesting() {
+  blink::WebRtcAudioSink* GetLocalTrackAudioSinkForTesting() {
     return local_track_audio_sink_.get();
   }
   MediaStreamVideoWebRtcSink* GetLocalTrackVideoSinkForTesting() {
@@ -126,7 +126,7 @@ class CONTENT_EXPORT WebRtcMediaStreamTrackAdapter
   scoped_refptr<webrtc::MediaStreamTrackInterface> webrtc_track_;
   // If the track is local, a sink is added to the local webrtc track that is
   // owned by us.
-  std::unique_ptr<WebRtcAudioSink> local_track_audio_sink_;
+  std::unique_ptr<blink::WebRtcAudioSink> local_track_audio_sink_;
   std::unique_ptr<MediaStreamVideoWebRtcSink> local_track_video_sink_;
   // If the track is remote, an adapter is used that listens to notifications on
   // the remote webrtc track and notifies Blink.

third_party/blink/public/BUILD.gn

@@ -161,6 +161,7 @@ source_set("blink_headers") {
     "platform/modules/peerconnection/two_keys_adapter_map.h",
     "platform/modules/peerconnection/web_rtc_video_encoder_factory.h",
     "platform/modules/peerconnection/web_rtc_video_frame_adapter_factory.h",
+    "platform/modules/peerconnection/webrtc_audio_sink.h",
     "platform/modules/peerconnection/webrtc_util.h",
     "platform/modules/remoteplayback/web_remote_playback_client.h",
     "platform/modules/service_worker/web_service_worker_error.h",
@@ -561,6 +562,7 @@ source_set("blink_headers") {
     "//third_party/webrtc/media:rtc_media_base",
     "//third_party/webrtc/modules/audio_device:audio_device_api",
     "//third_party/webrtc/modules/audio_processing:api",
+    "//third_party/webrtc/pc:peerconnection",
     "//third_party/webrtc/rtc_base:rtc_base",
     "//third_party/webrtc/rtc_base:rtc_task_queue",
     "//third_party/webrtc_overrides:init_webrtc",

third_party/blink/public/platform/modules/peerconnection/DEPS

 include_rules = [
     "+media/base/video_codecs.h",
+    "+media/base/audio_parameters.h",
+    "+media/base/audio_push_fifo.h",
 ]

content/renderer/media/webrtc/webrtc_audio_sink.h → third_party/blink/public/platform/modules/peerconnection/webrtc_audio_sink.h

@@ -2,8 +2,8 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_
+#ifndef THIRD_PARTY_BLINK_PUBLIC_PLATFORM_MODULES_PEERCONNECTION_WEBRTC_AUDIO_SINK_H_
+#define THIRD_PARTY_BLINK_PUBLIC_PLATFORM_MODULES_PEERCONNECTION_WEBRTC_AUDIO_SINK_H_
 
 #include <stdint.h>
 
@@ -17,17 +17,17 @@
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
-#include "content/common/content_export.h"
 #include "media/base/audio_parameters.h"
 #include "media/base/audio_push_fifo.h"
 #include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_level_calculator.h"
 #include "third_party/blink/public/platform/modules/mediastream/web_media_stream_audio_sink.h"
+#include "third_party/blink/public/platform/web_common.h"
 #include "third_party/webrtc/api/media_stream_interface.h"
 #include "third_party/webrtc/pc/media_stream_track.h"
 
-namespace content {
+namespace blink {
 
-// Provides an implementation of the blink::WebMediaStreamAudioSink which
+// Provides an implementation of the WebMediaStreamAudioSink which
 // re-chunks audio data into the 10ms chunks required by WebRTC and then
 // delivers the audio to one or more objects implementing the
 // webrtc::AudioTrackSinkInterface.
@@ -36,7 +36,11 @@ namespace content {
 // manages one or more "WebRTC sinks" (i.e., instances of
 // webrtc::AudioTrackSinkInterface) which are added/removed on the WebRTC
 // signaling thread.
-class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
+//
+// TODO(crbug.com/787254): Move this class out of the Blink exposed API when its
+// clients get Onion soup'ed. Also, switch away from using std::string and
+// std::vector.
+class BLINK_PLATFORM_EXPORT WebRtcAudioSink : public WebMediaStreamAudioSink {
  public:
   WebRtcAudioSink(
       const std::string& label,
@@ -54,8 +58,7 @@ class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
   // level. This is passed via the Adapter to libjingle. This method may only
   // be called once, before the audio data flow starts, and before any calls to
   // Adapter::GetSignalLevel() might be made.
-  void SetLevel(
-      scoped_refptr<blink::MediaStreamAudioLevelCalculator::Level> level);
+  void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
 
   // Set the processor that applies signal processing on the data from the
   // source. This is passed via the Adapter to libjingle. This method may only
@@ -89,7 +92,7 @@ class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
       audio_processor_ = std::move(processor);
     }
     void set_level(
-        scoped_refptr<blink::MediaStreamAudioLevelCalculator::Level> level) {
+        scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
       level_ = std::move(level);
     }
 
@@ -136,7 +139,7 @@ class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
     // Thread-safe accessor to current audio signal level. This may be null, if
     // not applicable to the current use case. This must be set before calls to
     // GetSignalLevel() are made.
-    scoped_refptr<blink::MediaStreamAudioLevelCalculator::Level> level_;
+    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
 
     // Lock that protects concurrent access to the |sinks_| list.
     base::Lock lock_;
@@ -148,7 +151,7 @@ class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
     DISALLOW_COPY_AND_ASSIGN(Adapter);
   };
 
-  // blink::WebMediaStreamAudioSink implementation.
+  // WebMediaStreamAudioSink implementation.
   void OnData(const media::AudioBus& audio_bus,
               base::TimeTicks estimated_capture_time) override;
   void OnSetFormat(const media::AudioParameters& params) override;
@@ -180,6 +183,6 @@ class CONTENT_EXPORT WebRtcAudioSink : public blink::WebMediaStreamAudioSink {
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioSink);
 };
 
-}  // namespace content
+}  // namespace blink
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_H_
+#endif  // THIRD_PARTY_BLINK_PUBLIC_PLATFORM_MODULES_PEERCONNECTION_WEBRTC_AUDIO_SINK_H_

third_party/blink/renderer/platform/BUILD.gn

@@ -1243,6 +1243,7 @@ jumbo_component("platform") {
     "peerconnection/rtc_video_encoder.cc",
     "peerconnection/rtc_video_encoder.h",
     "peerconnection/rtc_void_request.h",
+    "peerconnection/webrtc_audio_sink.cc",
     "prerender.cc",
     "prerender.h",
     "prerender_client.h",

content/renderer/media/webrtc/webrtc_audio_sink.cc → third_party/blink/renderer/platform/peerconnection/webrtc_audio_sink.cc

@@ -2,7 +2,7 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
+#include "third_party/blink/public/platform/modules/peerconnection/webrtc_audio_sink.h"
 
 #include <algorithm>
 #include <limits>
@@ -12,10 +12,9 @@
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/stl_util.h"
-#include "base/threading/thread_task_runner_handle.h"
 #include "third_party/webrtc/rtc_base/ref_counted_object.h"
 
-namespace content {
+namespace blink {
 
 WebRtcAudioSink::WebRtcAudioSink(
     const std::string& label,
@@ -45,7 +44,7 @@ void WebRtcAudioSink::SetAudioProcessor(
 }
 
 void WebRtcAudioSink::SetLevel(
-    scoped_refptr<blink::MediaStreamAudioLevelCalculator::Level> level) {
+    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
   DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
   DCHECK(level.get());
   adapter_->set_level(std::move(level));
@@ -93,8 +92,7 @@ void WebRtcAudioSink::DeliverRebufferedAudio(const media::AudioBus& audio_bus,
   audio_bus.ToInterleaved<media::SignedInt16SampleTypeTraits>(
       audio_bus.frames(), interleaved_data_.get());
   adapter_->DeliverPCMToWebRtcSinks(interleaved_data_.get(),
-                                    params_.sample_rate(),
-                                    audio_bus.channels(),
+                                    params_.sample_rate(), audio_bus.channels(),
                                     audio_bus.frames());
 }
 
@@ -144,8 +142,8 @@ std::string WebRtcAudioSink::Adapter::kind() const {
 bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksInCurrentSequence());
-  return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
-      set_enabled(enable);
+  return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::set_enabled(
+      enable);
 }
 
 void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
@@ -197,4 +195,4 @@ webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const {
   return source_.get();
 }
 
-}  // namespace content
+}  // namespace blink