MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_audio_capturer_unittest.cc

-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "base/logging.h"
-#include "build/build_config.h"
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/mock_media_constraint_factory.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/audio_bus.h"
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-
-using ::testing::_;
-using ::testing::AtLeast;
-
-namespace content {
-
-namespace {
-
-class MockCapturerSource : public media::AudioCapturerSource {
- public:
-  MockCapturerSource() {}
-  MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
-                                CaptureCallback* callback,
-                                int session_id));
-  MOCK_METHOD0(Start, void());
-  MOCK_METHOD0(Stop, void());
-  MOCK_METHOD1(SetVolume, void(double volume));
-  MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
-
- protected:
-  ~MockCapturerSource() override {}
-};
-
-class MockMediaStreamAudioSink : public MediaStreamAudioSink {
- public:
-  MockMediaStreamAudioSink() {}
-  ~MockMediaStreamAudioSink() override {}
-  void OnData(const media::AudioBus& audio_bus,
-              base::TimeTicks estimated_capture_time) override {
-    EXPECT_EQ(audio_bus.channels(), params_.channels());
-    EXPECT_EQ(audio_bus.frames(), params_.frames_per_buffer());
-    EXPECT_FALSE(estimated_capture_time.is_null());
-    OnDataCallback();
-  }
-  MOCK_METHOD0(OnDataCallback, void());
-  void OnSetFormat(const media::AudioParameters& params) override {
-    params_ = params;
-    FormatIsSet();
-  }
-  MOCK_METHOD0(FormatIsSet, void());
-
- private:
-  media::AudioParameters params_;
-};
-
-}  // namespace
-
-class WebRtcAudioCapturerTest : public testing::Test {
- protected:
-  WebRtcAudioCapturerTest()
-#if defined(OS_ANDROID)
-      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
-    // Android works with a buffer size bigger than 20ms.
-#else
-      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
-#endif
-  }
-
-  void VerifyAudioParams(const blink::WebMediaConstraints& constraints,
-                         bool need_audio_processing) {
-    capturer_ = WebRtcAudioCapturer::CreateCapturer(
-        -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
-                             params_.sample_rate(), params_.channel_layout(),
-                             params_.frames_per_buffer()),
-        constraints, NULL, NULL);
-    capturer_source_ = new MockCapturerSource();
-    EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1));
-    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
-    EXPECT_CALL(*capturer_source_.get(), Start());
-    capturer_->SetCapturerSource(capturer_source_, params_);
-
-    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-    track_.reset(new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
-    track_->Start();
-
-    // Connect a mock sink to the track.
-    scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
-    track_->AddSink(sink.get());
-
-    int delay_ms = 65;
-    bool key_pressed = true;
-    double volume = 0.9;
-
-    scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_);
-    audio_bus->Zero();
-
-    media::AudioCapturerSource::CaptureCallback* callback =
-        static_cast<media::AudioCapturerSource::CaptureCallback*>(
-            capturer_.get());
-
-    // Verify the sink is getting the correct values.
-    EXPECT_CALL(*sink, FormatIsSet());
-    EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1));
-    callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
-
-    track_->RemoveSink(sink.get());
-    EXPECT_CALL(*capturer_source_.get(), Stop());
-    capturer_->Stop();
-  }
-
-  media::AudioParameters params_;
-  scoped_refptr<MockCapturerSource> capturer_source_;
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
-  scoped_ptr<WebRtcLocalAudioTrack> track_;
-};
-
-TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) {
-  // Turn off the default constraints to verify that the sink will get packets
-  // with a buffer size smaller than 10ms.
-  MockMediaConstraintFactory constraint_factory;
-  constraint_factory.DisableDefaultAudioConstraints();
-  VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false);
-}
-
-TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) {
-  MockMediaConstraintFactory constraint_factory;
-  const std::string dummy_constraint = "dummy";
-  constraint_factory.AddMandatory(dummy_constraint, true);
-
-  scoped_refptr<WebRtcAudioCapturer> capturer(
-      WebRtcAudioCapturer::CreateCapturer(
-          0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
-                              params_.sample_rate(), params_.channel_layout(),
-                              params_.frames_per_buffer()),
-          constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
-  EXPECT_TRUE(capturer.get() == NULL);
-}
-
-
-}  // namespace content