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// Copyright 2020 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/platform/peerconnection/webrtc_audio_sink.h"
#include "base/memory/raw_ptr.h"
#include "media/base/fake_single_thread_task_runner.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
using testing::_;
namespace blink {
namespace {
class MockAudioSink : public webrtc::AudioTrackSinkInterface {
public:
MockAudioSink() = default;
~MockAudioSink() override = default;
MOCK_METHOD6(OnData,
void(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_samples,
std::optional<int64_t> absolute_capture_timestamp_ms));
};
} // namespace
TEST(WebRtcAudioSinkTest, CaptureTimestamp) {
MockAudioSink sink_1;
MockAudioSink sink_2;
base::SimpleTestTickClock dummy_clock;
std::unique_ptr<WebRtcAudioSink> webrtc_audio_sink(
new WebRtcAudioSink("test_sink", nullptr,
/*signaling_task_runner=*/
new media::FakeSingleThreadTaskRunner(&dummy_clock),
/*main_task_runner=*/
new media::FakeSingleThreadTaskRunner(&dummy_clock)));
// |web_media_stream_audio_sink| is to access methods that are privately
// inherited by WebRtcAudioSink.
WebMediaStreamAudioSink* const web_media_stream_audio_sink =
static_cast<WebMediaStreamAudioSink*>(webrtc_audio_sink.get());
webrtc_audio_sink->webrtc_audio_track()->AddSink(&sink_1);
webrtc_audio_sink->webrtc_audio_track()->AddSink(&sink_2);
constexpr int kInputChannels = 2;
constexpr int kInputFramesPerBuffer = 96;
constexpr int kSampleRateHz = 8000;
constexpr int kOutputFramesPerBuffer = kSampleRateHz / 100;
constexpr int kEnqueueFrames = kInputFramesPerBuffer - kOutputFramesPerBuffer;
constexpr int64_t kStartCaptureTimestampMs = 12345678;
constexpr int64_t kCaptureIntervalMs = 567;
web_media_stream_audio_sink->OnSetFormat(
media::AudioParameters(media::AudioParameters::AUDIO_PCM_LINEAR,
media::ChannelLayoutConfig::Stereo(),
kSampleRateHz, kOutputFramesPerBuffer));
std::unique_ptr<media::AudioBus> bus =
media::AudioBus::Create(kInputChannels, kInputFramesPerBuffer);
bus->Zero();
base::TimeTicks capture_time =
base::TimeTicks() + base::Milliseconds(kStartCaptureTimestampMs);
{
const int64_t expected_capture_time_ms =
(capture_time - base::TimeTicks()).InMilliseconds();
EXPECT_CALL(
sink_1,
OnData(_, _, kSampleRateHz, kInputChannels, kOutputFramesPerBuffer,
std::make_optional<int64_t>(expected_capture_time_ms)))
.Times(1);
EXPECT_CALL(
sink_2,
OnData(_, _, kSampleRateHz, kInputChannels, kOutputFramesPerBuffer,
std::make_optional<int64_t>(expected_capture_time_ms)))
.Times(1);
web_media_stream_audio_sink->OnData(*bus, capture_time);
}
capture_time += base::Milliseconds(kCaptureIntervalMs);
{
const int64_t expected_capture_time_ms =
(capture_time - base::TimeTicks()).InMilliseconds() -
((kEnqueueFrames * 1000) / kSampleRateHz);
EXPECT_CALL(
sink_1,
OnData(_, _, kSampleRateHz, kInputChannels, kOutputFramesPerBuffer,
std::make_optional<int64_t>(expected_capture_time_ms)))
.Times(1);
EXPECT_CALL(
sink_2,
OnData(_, _, kSampleRateHz, kInputChannels, kOutputFramesPerBuffer,
std::make_optional<int64_t>(expected_capture_time_ms)))
.Times(1);
web_media_stream_audio_sink->OnData(*bus, capture_time);
}
}
} // namespace blink
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