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3e70781361
This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
114 lines
3.6 KiB
C++
114 lines
3.6 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 < AudioFrame::kMaxDataSizeSamples; 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 kNumChannels = 1;
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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 < AudioFrame::kMaxDataSizeSamples; 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, UpdateFrame) {
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AudioFrame frame;
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int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
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frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
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AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
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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(kNumChannels, frame.num_channels_);
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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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kNumChannels);
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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, CopyFrom) {
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AudioFrame frame1;
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AudioFrame frame2;
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int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
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frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
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AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
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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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kNumChannels);
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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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