2250b05778
Two new classes are added to WebRTC from Chrome: ChannelMixer and ChannelMixingMatrix but they are not yet utilized in the audio path for WebRTC. The idea is to utilize these new classes when adding support for multi- channel encoding/decoding in WebRTC/Chrome. Adds support for a new enumerator call webrtc::ChannelLayout and some helper methods which maps between channel layout and number of channels. These parts are also copied from Chrome. Minor (cosmetic) changes are also done on the AudioFrame to prepare for upcoming work. Bug: webrtc:10783 Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28482}
136 lines
4.6 KiB
C++
136 lines
4.6 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdint.h>
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#include <string.h> // memcmp
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#include "api/audio/audio_frame.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
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const int16_t* frame_data = frame.data();
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for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
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if (frame_data[i] != sample) {
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return false;
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}
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}
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return true;
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}
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constexpr uint32_t kTimestamp = 27;
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constexpr int kSampleRateHz = 16000;
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constexpr size_t kNumChannelsMono = 1;
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constexpr size_t kNumChannelsStereo = 2;
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constexpr size_t kNumChannels5_1 = 6;
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constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
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} // namespace
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TEST(AudioFrameTest, FrameStartsMuted) {
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AudioFrame frame;
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EXPECT_TRUE(frame.muted());
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EXPECT_TRUE(AllSamplesAre(0, frame));
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}
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TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
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AudioFrame frame;
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frame.mutable_data();
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EXPECT_FALSE(frame.muted());
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EXPECT_TRUE(AllSamplesAre(0, frame));
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}
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TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
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AudioFrame frame;
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int16_t* frame_data = frame.mutable_data();
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for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
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frame_data[i] = 17;
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}
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ASSERT_TRUE(AllSamplesAre(17, frame));
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frame.Mute();
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EXPECT_TRUE(frame.muted());
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EXPECT_TRUE(AllSamplesAre(0, frame));
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}
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TEST(AudioFrameTest, UpdateFrameMono) {
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AudioFrame frame;
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int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17};
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frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
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AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannelsMono);
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EXPECT_EQ(kTimestamp, frame.timestamp_);
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EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
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EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz());
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EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
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EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
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EXPECT_EQ(kNumChannelsMono, frame.num_channels());
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EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame.channel_layout());
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EXPECT_FALSE(frame.muted());
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EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
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frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
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kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
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kNumChannelsMono);
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EXPECT_TRUE(frame.muted());
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EXPECT_TRUE(AllSamplesAre(0, frame));
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}
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TEST(AudioFrameTest, UpdateFrameMultiChannel) {
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AudioFrame frame;
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frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
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kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
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kNumChannelsStereo);
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EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
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EXPECT_EQ(kNumChannelsStereo, frame.num_channels());
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EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame.channel_layout());
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frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
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kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
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kNumChannels5_1);
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EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
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EXPECT_EQ(kNumChannels5_1, frame.num_channels());
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EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame.channel_layout());
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}
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TEST(AudioFrameTest, CopyFrom) {
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AudioFrame frame1;
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AudioFrame frame2;
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int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17};
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frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
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AudioFrame::kPLC, AudioFrame::kVadActive,
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kNumChannelsMono);
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frame1.CopyFrom(frame2);
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EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
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EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
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EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
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EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
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EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
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EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
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EXPECT_EQ(frame2.muted(), frame1.muted());
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EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
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frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
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kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
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kNumChannelsMono);
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frame1.CopyFrom(frame2);
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EXPECT_EQ(frame2.muted(), frame1.muted());
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EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
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}
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} // namespace webrtc
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