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Unified Diff: content/renderer/media/webrtc_audio_capturer_unittest.cc

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
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Index: content/renderer/media/webrtc_audio_capturer_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc
deleted file mode 100644
index 306ca9882d773e0e80188b1b052a290482f209d1..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc
+++ /dev/null
@@ -1,149 +0,0 @@
-// 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
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