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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
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 "api/audio/audio_frame.h"
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#include <stdint.h>
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#include <string.h> // memcmp
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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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