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// Copyright 2016 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 <memory>
#include <string>
#include "base/bind.h"
#include "build/build_config.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_parameters.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "third_party/blink/public/platform/modules/mediastream/web_media_stream_audio_sink.h"
#include "third_party/blink/public/platform/scheduler/test/renderer_scheduler_test_support.h"
#include "third_party/blink/public/web/web_heap.h"
#include "third_party/blink/renderer/modules/mediastream/processed_local_audio_source.h"
#include "third_party/blink/renderer/modules/mediastream/testing_platform_support_with_mock_audio_capture_source.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h"
#include "third_party/blink/renderer/platform/mediastream/media_constraints.h"
#include "third_party/blink/renderer/platform/mediastream/media_stream_audio_processor_options.h"
#include "third_party/blink/renderer/platform/mediastream/media_stream_audio_track.h"
using ::testing::_;
using ::testing::AtLeast;
using ::testing::Invoke;
using ::testing::WithArg;
namespace blink {
namespace {
// Audio parameters for the VerifyAudioFlowWithoutAudioProcessing test.
constexpr int kSampleRate = 48000;
constexpr media::ChannelLayout kChannelLayout = media::CHANNEL_LAYOUT_STEREO;
constexpr int kRequestedBufferSize = 512;
// On Android, ProcessedLocalAudioSource forces a 20ms buffer size from the
// input device.
#if defined(OS_ANDROID)
constexpr int kExpectedSourceBufferSize = kSampleRate / 50;
#else
constexpr int kExpectedSourceBufferSize = kRequestedBufferSize;
#endif
// On both platforms, even though audio processing is turned off, the
// MediaStreamAudioProcessor will force the use of 10ms buffer sizes on the
// output end of its FIFO.
constexpr int kExpectedOutputBufferSize = kSampleRate / 100;
class MockMediaStreamAudioSink : public WebMediaStreamAudioSink {
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(params_);
}
MOCK_METHOD1(FormatIsSet, void(const media::AudioParameters& params));
private:
media::AudioParameters params_;
};
} // namespace
class ProcessedLocalAudioSourceTest : public testing::Test {
protected:
ProcessedLocalAudioSourceTest() {}
~ProcessedLocalAudioSourceTest() override {}
void SetUp() override {
blink_audio_source_.Initialize(blink::WebString::FromUTF8("audio_label"),
WebMediaStreamSource::kTypeAudio,
WebString::FromUTF8("audio_track"),
false /* remote */);
blink_audio_track_.Initialize(blink_audio_source_.Id(),
blink_audio_source_);
}
void TearDown() override {
blink_audio_track_.Reset();
blink_audio_source_.Reset();
WebHeap::CollectAllGarbageForTesting();
}
void CreateProcessedLocalAudioSource(
const AudioProcessingProperties& properties) {
std::unique_ptr<blink::ProcessedLocalAudioSource> source =
std::make_unique<blink::ProcessedLocalAudioSource>(
nullptr /* consumer_web_frame is N/A for non-browser tests */,
MediaStreamDevice(mojom::MediaStreamType::DEVICE_AUDIO_CAPTURE,
"mock_audio_device_id", "Mock audio device",
kSampleRate, kChannelLayout,
kRequestedBufferSize),
false /* disable_local_echo */, properties, base::DoNothing(),
scheduler::GetSingleThreadTaskRunnerForTesting());
source->SetAllowInvalidRenderFrameIdForTesting(true);
blink_audio_source_.SetPlatformSource(
std::move(source)); // Takes ownership.
}
void CheckSourceFormatMatches(const media::AudioParameters& params) {
EXPECT_EQ(kSampleRate, params.sample_rate());
EXPECT_EQ(kChannelLayout, params.channel_layout());
EXPECT_EQ(kExpectedSourceBufferSize, params.frames_per_buffer());
}
void CheckOutputFormatMatches(const media::AudioParameters& params) {
EXPECT_EQ(kSampleRate, params.sample_rate());
EXPECT_EQ(kChannelLayout, params.channel_layout());
EXPECT_EQ(kExpectedOutputBufferSize, params.frames_per_buffer());
}
media::AudioCapturerSource::CaptureCallback* capture_source_callback() const {
return static_cast<media::AudioCapturerSource::CaptureCallback*>(
ProcessedLocalAudioSource::From(audio_source()));
}
MediaStreamAudioSource* audio_source() const {
return MediaStreamAudioSource::From(blink_audio_source_);
}
const WebMediaStreamTrack& blink_audio_track() { return blink_audio_track_; }
MockAudioCapturerSource* mock_audio_capturer_source() {
return webrtc_audio_device_platform_support_->mock_audio_capturer_source();
}
private:
ScopedTestingPlatformSupport<AudioCapturerSourceTestingPlatformSupport>
webrtc_audio_device_platform_support_;
WebMediaStreamSource blink_audio_source_;
WebMediaStreamTrack blink_audio_track_;
};
// Tests a basic end-to-end start-up, track+sink connections, audio flow, and
// shut-down. The unit tests in media_stream_audio_unittest.cc provide more
// comprehensive testing of the object graph connections and multi-threading
// concerns.
TEST_F(ProcessedLocalAudioSourceTest, VerifyAudioFlowWithoutAudioProcessing) {
using ThisTest =
ProcessedLocalAudioSourceTest_VerifyAudioFlowWithoutAudioProcessing_Test;
// Turn off the default constraints so the sink will get audio in chunks of
// the native buffer size.
AudioProcessingProperties properties;
properties.DisableDefaultProperties();
CreateProcessedLocalAudioSource(properties);
// Connect the track, and expect the MockAudioCapturerSource to be initialized
// and started by ProcessedLocalAudioSource.
EXPECT_CALL(*mock_audio_capturer_source(),
Initialize(_, capture_source_callback()))
.WillOnce(WithArg<0>(Invoke(this, &ThisTest::CheckSourceFormatMatches)));
EXPECT_CALL(*mock_audio_capturer_source(), SetAutomaticGainControl(true));
EXPECT_CALL(*mock_audio_capturer_source(), Start())
.WillOnce(Invoke(
capture_source_callback(),
&media::AudioCapturerSource::CaptureCallback::OnCaptureStarted));
ASSERT_TRUE(audio_source()->ConnectToTrack(blink_audio_track()));
CheckOutputFormatMatches(audio_source()->GetAudioParameters());
// Connect a sink to the track.
std::unique_ptr<MockMediaStreamAudioSink> sink(
new MockMediaStreamAudioSink());
EXPECT_CALL(*sink, FormatIsSet(_))
.WillOnce(Invoke(this, &ThisTest::CheckOutputFormatMatches));
MediaStreamAudioTrack::From(blink_audio_track())->AddSink(sink.get());
// Feed audio data into the ProcessedLocalAudioSource and expect it to reach
// the sink.
int delay_ms = 65;
bool key_pressed = true;
double volume = 0.9;
const base::TimeTicks capture_time =
base::TimeTicks::Now() + base::TimeDelta::FromMilliseconds(delay_ms);
std::unique_ptr<media::AudioBus> audio_bus =
media::AudioBus::Create(2, kExpectedSourceBufferSize);
audio_bus->Zero();
EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1));
capture_source_callback()->Capture(audio_bus.get(), capture_time, volume,
key_pressed);
// Expect the ProcessedLocalAudioSource to auto-stop the MockCapturerSource
// when the track is stopped.
EXPECT_CALL(*mock_audio_capturer_source(), Stop());
MediaStreamAudioTrack::From(blink_audio_track())->Stop();
}
} // namespace blink
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