fca90714b3
This is a preparation to fix each files. This CL has no behavior changes. This patch was fully automated using script: https://paste.googleplex.com/5614491201175552 See internal doc about it: https://docs.google.com/document/d/1erdcokeh6rfBqs_h0drHqSLtbDbB61j7j3O2Pz8NH78/edit?resourcekey=0-hNe6w1hYAYyVXGEpWI7HVA&tab=t.0 Bug: 40285824 Change-Id: Ibfa3bf8ef4117b3c6c73d64c1295368ecdf16c18 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5759698 Reviewed-by: Joe Downing <joedow@chromium.org> Commit-Queue: Joe Downing <joedow@chromium.org> Auto-Submit: Arthur Sonzogni <arthursonzogni@chromium.org> Cr-Commit-Position: refs/heads/main@{#1344112}
124 lines
4.2 KiB
C++
124 lines
4.2 KiB
C++
// Copyright 2016 The Chromium Authors
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// Use of this source code is governed by a BSD-style license that can be
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// found in the LICENSE file.
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#ifdef UNSAFE_BUFFERS_BUILD
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// TODO(crbug.com/40285824): Remove this and convert code to safer constructs.
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#pragma allow_unsafe_buffers
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#endif
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#include "remoting/protocol/webrtc_audio_source_adapter.h"
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#include <numeric>
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#include <vector>
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#include "base/memory/ptr_util.h"
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#include "base/memory/raw_ptr.h"
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#include "base/run_loop.h"
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#include "base/test/task_environment.h"
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#include "remoting/proto/audio.pb.h"
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#include "remoting/protocol/fake_audio_source.h"
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "third_party/webrtc/api/media_stream_interface.h"
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#include "third_party/webrtc/rtc_base/ref_count.h"
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#include "third_party/webrtc/rtc_base/ref_counted_object.h"
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namespace remoting::protocol {
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namespace {
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const int kSampleRate = 48000;
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const int kBytesPerSample = 2;
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const int kChannels = 2;
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constexpr auto kFrameDuration = base::Milliseconds(10);
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class FakeAudioSink : public webrtc::AudioTrackSinkInterface {
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public:
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FakeAudioSink() = default;
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~FakeAudioSink() override = default;
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void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_samples) override {
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EXPECT_EQ(kSampleRate, sample_rate);
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EXPECT_EQ(kBytesPerSample * 8, bits_per_sample);
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EXPECT_EQ(kChannels, static_cast<int>(number_of_channels));
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EXPECT_EQ((kSampleRate * kFrameDuration).InSeconds(),
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static_cast<int>(number_of_samples));
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const int16_t* samples = reinterpret_cast<const int16_t*>(audio_data);
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samples_.insert(samples_.end(), samples,
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samples + number_of_samples * kChannels);
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}
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const std::vector<int16_t>& samples() { return samples_; }
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private:
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std::vector<int16_t> samples_;
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};
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} // namespace
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class WebrtcAudioSourceAdapterTest : public testing::Test {
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public:
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void SetUp() override {
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audio_source_adapter_ = new rtc::RefCountedObject<WebrtcAudioSourceAdapter>(
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task_environment_.GetMainThreadTaskRunner());
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audio_source_ = new FakeAudioSource();
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audio_source_adapter_->Start(base::WrapUnique(audio_source_.get()));
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audio_source_adapter_->AddSink(&sink_);
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base::RunLoop().RunUntilIdle();
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}
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void TearDown() override {
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audio_source_adapter_ = nullptr;
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base::RunLoop().RunUntilIdle();
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}
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protected:
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base::test::SingleThreadTaskEnvironment task_environment_;
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raw_ptr<FakeAudioSource, AcrossTasksDanglingUntriaged> audio_source_;
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scoped_refptr<WebrtcAudioSourceAdapter> audio_source_adapter_;
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FakeAudioSink sink_;
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};
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TEST_F(WebrtcAudioSourceAdapterTest, PartialFrames) {
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int16_t sample_value = 1;
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std::vector<int> frame_sizes_ms = {10, 12, 18, 2, 5, 7, 55, 13, 8};
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for (int frame_size_ms : frame_sizes_ms) {
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int num_samples = frame_size_ms * kSampleRate / 1000;
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std::vector<int16_t> data(num_samples * kChannels);
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for (int i = 0; i < num_samples; ++i) {
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data[i * kChannels] = sample_value;
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data[i * kChannels + 1] = -sample_value;
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++sample_value;
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}
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std::unique_ptr<AudioPacket> packet(new AudioPacket());
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packet->add_data(reinterpret_cast<char*>(&(data[0])),
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num_samples * kChannels * sizeof(int16_t));
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packet->set_encoding(AudioPacket::ENCODING_RAW);
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packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000);
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packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
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packet->set_channels(AudioPacket::CHANNELS_STEREO);
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audio_source_->callback().Run(std::move(packet));
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}
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int total_length_ms =
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std::accumulate(frame_sizes_ms.begin(), frame_sizes_ms.end(), 0,
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[](int sum, int x) { return sum + x; });
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const std::vector<int16_t>& received = sink_.samples();
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int total_samples = total_length_ms * kSampleRate / 1000;
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ASSERT_EQ(total_samples * kChannels, static_cast<int>(received.size()));
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sample_value = 1;
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for (int i = 0; i < total_samples; ++i) {
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ASSERT_EQ(sample_value, received[i * kChannels]) << i;
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ASSERT_EQ(-sample_value, received[i * kChannels + 1]);
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++sample_value;
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}
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}
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} // namespace remoting::protocol
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