Rename WebRtcLocalAudioSourceProviderTest to WebAudioMediaStreamAudioSinkTest

... as it is not specific to webrtc.

BUG=577874,704136
R=guidou@chromium.org, haraken@chromium.org

Change-Id: I4357d66c198acb2911a6ea3ec1e65bb443ca23c2
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1661609
Auto-Submit: Antonio Gomes <tonikitoo@igalia.com>
Commit-Queue: Kentaro Hara <haraken@chromium.org>
Reviewed-by: Kentaro Hara <haraken@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#669977}

content/renderer/media/stream/media_stream_center.cc

@@ -26,7 +26,7 @@
 #include "third_party/blink/public/platform/web_vector.h"
 #include "third_party/blink/public/web/modules/mediastream/media_stream_video_source.h"
 #include "third_party/blink/public/web/modules/mediastream/media_stream_video_track.h"
-#include "third_party/blink/public/web/modules/mediastream/webrtc_local_audio_source_provider.h"
+#include "third_party/blink/public/web/modules/mediastream/webaudio_media_stream_audio_sink.h"
 #include "third_party/blink/public/web/web_frame.h"
 
 using blink::WebFrame;
@@ -194,10 +194,7 @@ MediaStreamCenter::CreateWebAudioSourceFromMediaStreamTrack(
   blink::WebMediaStreamSource source = track.Source();
   DCHECK_EQ(source.GetType(), blink::WebMediaStreamSource::kTypeAudio);
 
-  // TODO(tommi): Rename WebRtcLocalAudioSourceProvider to
-  // WebAudioMediaStreamSink since it's not specific to any particular source.
-  // https://crbug.com/577874
-  return new blink::WebRtcLocalAudioSourceProvider(track, context_sample_rate);
+  return new blink::WebAudioMediaStreamAudioSink(track, context_sample_rate);
 }
 
 void MediaStreamCenter::DidStopMediaStreamSource(

third_party/blink/public/BUILD.gn

@@ -379,7 +379,7 @@ source_set("blink_headers") {
     "web/modules/mediastream/media_stream_video_track.h",
     "web/modules/mediastream/video_track_adapter_settings.h",
     "web/modules/mediastream/web_media_stream_utils.h",
-    "web/modules/mediastream/webrtc_local_audio_source_provider.h",
+    "web/modules/mediastream/webaudio_media_stream_audio_sink.h",
     "web/modules/service_worker/web_service_worker_context_client.h",
     "web/modules/service_worker/web_service_worker_context_proxy.h",
     "web/web_active_fling_parameters.h",

third_party/blink/public/web/modules/mediastream/webrtc_local_audio_source_provider.h → third_party/blink/public/web/modules/mediastream/webaudio_media_stream_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 THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
-#define THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
+#ifndef THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBAUDIO_MEDIA_STREAM_AUDIO_SINK_H_
+#define THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBAUDIO_MEDIA_STREAM_AUDIO_SINK_H_
 
 #include <stddef.h>
 
@@ -32,14 +32,10 @@ namespace blink {
 
 class WebAudioSourceProviderClient;
 
-// TODO(miu): This implementation should be renamed to WebAudioMediaStreamSink,
-// as it should work as a provider for WebAudio from ANY MediaStreamAudioTrack.
-// http://crbug.com/577874
-//
-// WebRtcLocalAudioSourceProvider provides a bridge between classes:
+// WebAudioMediaStreamAudioSink provides a bridge between classes:
 //     MediaStreamAudioTrack ---> WebAudioSourceProvider
 //
-// WebRtcLocalAudioSourceProvider works as a sink to the MediaStreamAudioTrack
+// WebAudioMediaStreamAudioSink works as a sink to the MediaStreamAudioTrack
 // and stores the capture data to a FIFO. When the media stream is connected to
 // WebAudio MediaStreamAudioSourceNode as a source provider,
 // MediaStreamAudioSourceNode will periodically call provideInput() to get the
@@ -49,16 +45,16 @@ class WebAudioSourceProviderClient;
 //
 // TODO(crbug.com/704136): Move this class out of the Blink exposed API
 // when all users of it have been Onion souped.
-class BLINK_MODULES_EXPORT WebRtcLocalAudioSourceProvider
+class BLINK_MODULES_EXPORT WebAudioMediaStreamAudioSink
     : public WebAudioSourceProvider,
       public media::AudioConverter::InputCallback,
       public WebMediaStreamAudioSink {
  public:
   static const size_t kWebAudioRenderBufferSize;
 
-  explicit WebRtcLocalAudioSourceProvider(const WebMediaStreamTrack& track,
-                                          int context_sample_rate);
-  ~WebRtcLocalAudioSourceProvider() override;
+  explicit WebAudioMediaStreamAudioSink(const WebMediaStreamTrack& track,
+                                        int context_sample_rate);
+  ~WebAudioMediaStreamAudioSink() override;
 
   // WebMediaStreamAudioSink implementation.
   void OnData(const media::AudioBus& audio_bus,
@@ -120,9 +116,9 @@ class BLINK_MODULES_EXPORT WebRtcLocalAudioSourceProvider
   // Used to assert that OnReadyStateChanged() is not accessed concurrently.
   REENTRANCY_CHECKER(ready_state_reentrancy_checker_);
 
-  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
+  DISALLOW_COPY_AND_ASSIGN(WebAudioMediaStreamAudioSink);
 };
 
 }  // namespace blink
 
-#endif  // THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
+#endif  // THIRD_PARTY_BLINK_PUBLIC_WEB_MODULES_MEDIASTREAM_WEBAUDIO_MEDIA_STREAM_AUDIO_SINK_H_

third_party/blink/renderer/modules/BUILD.gn

@@ -338,7 +338,7 @@ jumbo_source_set("unit_tests") {
     "mediastream/mock_mojo_media_stream_dispatcher_host.cc",
     "mediastream/mock_mojo_media_stream_dispatcher_host.h",
     "mediastream/video_track_adapter_unittest.cc",
-    "mediastream/webrtc_local_audio_source_provider_test.cc",
+    "mediastream/webaudio_media_stream_audio_sink_test.cc",
     "nfc/nfc_proxy_test.cc",
     "notifications/notification_data_test.cc",
     "notifications/notification_image_loader_test.cc",

third_party/blink/renderer/modules/mediastream/BUILD.gn

@@ -54,7 +54,7 @@ blink_modules_sources("mediastream") {
     "video_track_adapter.h",
     "video_track_adapter_settings.cc",
     "web_media_stream_utils.cc",
-    "webrtc_local_audio_source_provider.cc",
+    "webaudio_media_stream_audio_sink.cc",
   ]
 }
 

third_party/blink/renderer/modules/mediastream/webrtc_local_audio_source_provider.cc → third_party/blink/renderer/modules/mediastream/webaudio_media_stream_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 "third_party/blink/public/web/modules/mediastream/webrtc_local_audio_source_provider.h"
+#include "third_party/blink/public/web/modules/mediastream/webaudio_media_stream_audio_sink.h"
 
 #include <string>
 
@@ -21,9 +21,9 @@ namespace blink {
 // Size of the buffer that WebAudio processes each time, it is the same value
 // as AudioNode::ProcessingSizeInFrames in WebKit.
 // static
-const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
+const size_t WebAudioMediaStreamAudioSink::kWebAudioRenderBufferSize = 128;
 
-WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
+WebAudioMediaStreamAudioSink::WebAudioMediaStreamAudioSink(
     const WebMediaStreamTrack& track,
     int context_sample_rate)
     : is_enabled_(false), track_(track), track_stopped_(false) {
@@ -40,7 +40,7 @@ WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
   WebMediaStreamAudioSink::AddToAudioTrack(this, track_);
 }
 
-WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
+WebAudioMediaStreamAudioSink::~WebAudioMediaStreamAudioSink() {
   if (audio_converter_.get())
     audio_converter_->RemoveInput(this);
 
@@ -50,7 +50,7 @@ WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
     WebMediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
 }
 
-void WebRtcLocalAudioSourceProvider::OnSetFormat(
+void WebAudioMediaStreamAudioSink::OnSetFormat(
     const media::AudioParameters& params) {
   DCHECK(params.IsValid());
 
@@ -70,14 +70,14 @@ void WebRtcLocalAudioSourceProvider::OnSetFormat(
       kMaxNumberOfAudioFifoBuffers * params.frames_per_buffer()));
 }
 
-void WebRtcLocalAudioSourceProvider::OnReadyStateChanged(
+void WebAudioMediaStreamAudioSink::OnReadyStateChanged(
     WebMediaStreamSource::ReadyState state) {
   NON_REENTRANT_SCOPE(ready_state_reentrancy_checker_);
   if (state == WebMediaStreamSource::kReadyStateEnded)
     track_stopped_ = true;
 }
 
-void WebRtcLocalAudioSourceProvider::OnData(
+void WebAudioMediaStreamAudioSink::OnData(
     const media::AudioBus& audio_bus,
     base::TimeTicks estimated_capture_time) {
   NON_REENTRANT_SCOPE(capture_reentrancy_checker_);
@@ -100,12 +100,12 @@ void WebRtcLocalAudioSourceProvider::OnData(
   }
 }
 
-void WebRtcLocalAudioSourceProvider::SetClient(
+void WebAudioMediaStreamAudioSink::SetClient(
     WebAudioSourceProviderClient* client) {
   NOTREACHED();
 }
 
-void WebRtcLocalAudioSourceProvider::ProvideInput(
+void WebAudioMediaStreamAudioSink::ProvideInput(
     const WebVector<float*>& audio_data,
     size_t number_of_frames) {
   NON_REENTRANT_SCOPE(provide_input_reentrancy_checker_);
@@ -133,8 +133,8 @@ void WebRtcLocalAudioSourceProvider::ProvideInput(
 // AudioConverter which in turn is called by the above ProvideInput() function.
 // Thus thread safety analysis is disabled here and |lock_| acquire manually
 // asserted.
-double WebRtcLocalAudioSourceProvider::ProvideInput(media::AudioBus* audio_bus,
-                                                    uint32_t frames_delayed)
+double WebAudioMediaStreamAudioSink::ProvideInput(media::AudioBus* audio_bus,
+                                                  uint32_t frames_delayed)
     NO_THREAD_SAFETY_ANALYSIS {
   lock_.AssertAcquired();
   if (fifo_->frames() >= audio_bus->frames()) {
@@ -148,7 +148,7 @@ double WebRtcLocalAudioSourceProvider::ProvideInput(media::AudioBus* audio_bus,
   return 1.0;
 }
 
-void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
+void WebAudioMediaStreamAudioSink::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   base::AutoLock auto_lock(lock_);
   sink_params_ = sink_params;

third_party/blink/renderer/modules/mediastream/webrtc_local_audio_source_provider_test.cc → third_party/blink/renderer/modules/mediastream/webaudio_media_stream_audio_sink_test.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 "third_party/blink/public/web/modules/mediastream/webrtc_local_audio_source_provider.h"
+#include "third_party/blink/public/web/modules/mediastream/webaudio_media_stream_audio_sink.h"
 
 #include <stddef.h>
 
@@ -17,16 +17,15 @@
 
 namespace blink {
 
-class WebRtcLocalAudioSourceProviderTest : public testing::Test {
+class WebAudioMediaStreamAudioSinkTest : public testing::Test {
  protected:
   void SetUp() override {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 48000, 480);
     const int context_sample_rate = 44100;
-    sink_params_.Reset(
-        media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-        media::CHANNEL_LAYOUT_STEREO, context_sample_rate,
-        WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
+    sink_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                       media::CHANNEL_LAYOUT_STEREO, context_sample_rate,
+                       WebAudioMediaStreamAudioSink::kWebAudioRenderBufferSize);
     sink_bus_ = media::AudioBus::Create(sink_params_);
     WebMediaStreamSource audio_source;
     audio_source.Initialize(WebString::FromUTF8("dummy_source_id"),
@@ -37,7 +36,7 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
     blink_track_.SetPlatformTrack(
         std::make_unique<MediaStreamAudioTrack>(true));
     source_provider_.reset(
-        new WebRtcLocalAudioSourceProvider(blink_track_, context_sample_rate));
+        new WebAudioMediaStreamAudioSink(blink_track_, context_sample_rate));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }
@@ -52,10 +51,10 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
   media::AudioParameters sink_params_;
   std::unique_ptr<media::AudioBus> sink_bus_;
   WebMediaStreamTrack blink_track_;
-  std::unique_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
+  std::unique_ptr<WebAudioMediaStreamAudioSink> source_provider_;
 };
 
-TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
+TEST_F(WebAudioMediaStreamAudioSinkTest, VerifyDataFlow) {
   // TODO(miu): This test should be re-worked so that the audio data and format
   // is feed into a MediaStreamAudioSource and, through the
   // MediaStreamAudioTrack, ultimately delivered to the |source_provider_|.
@@ -114,7 +113,7 @@ TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
   }
 }
 
-TEST_F(WebRtcLocalAudioSourceProviderTest,
+TEST_F(WebAudioMediaStreamAudioSinkTest,
        DeleteSourceProviderBeforeStoppingTrack) {
   source_provider_.reset();
 
@@ -122,7 +121,7 @@ TEST_F(WebRtcLocalAudioSourceProviderTest,
   MediaStreamAudioTrack::From(blink_track_)->Stop();
 }
 
-TEST_F(WebRtcLocalAudioSourceProviderTest,
+TEST_F(WebAudioMediaStreamAudioSinkTest,
        StopTrackBeforeDeletingSourceProvider) {
   // Stop the audio track.
   MediaStreamAudioTrack::From(blink_track_)->Stop();