blob: b969d90a4c1a9e8286f96afb370c0663808485b6 [file] [log] [blame]
/*
* Copyright 2018 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string.h> // memcmp
#include "api/audio/audio_frame.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
const int16_t* frame_data = frame.data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
if (frame_data[i] != sample) {
return false;
}
}
return true;
}
constexpr uint32_t kTimestamp = 27;
constexpr int kSampleRateHz = 16000;
constexpr size_t kNumChannels = 1;
constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
} // namespace
TEST(AudioFrameTest, FrameStartsMuted) {
AudioFrame frame;
EXPECT_TRUE(frame.muted());
EXPECT_TRUE(AllSamplesAre(0, frame));
}
TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
AudioFrame frame;
frame.mutable_data();
EXPECT_FALSE(frame.muted());
EXPECT_TRUE(AllSamplesAre(0, frame));
}
TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
AudioFrame frame;
int16_t* frame_data = frame.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
frame_data[i] = 17;
}
ASSERT_TRUE(AllSamplesAre(17, frame));
frame.Mute();
EXPECT_TRUE(frame.muted());
EXPECT_TRUE(AllSamplesAre(0, frame));
}
TEST(AudioFrameTest, UpdateFrame) {
AudioFrame frame;
int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
EXPECT_EQ(kTimestamp, frame.timestamp_);
EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
EXPECT_EQ(kNumChannels, frame.num_channels_);
EXPECT_FALSE(frame.muted());
EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
kNumChannels);
EXPECT_TRUE(frame.muted());
EXPECT_TRUE(AllSamplesAre(0, frame));
}
TEST(AudioFrameTest, CopyFrom) {
AudioFrame frame1;
AudioFrame frame2;
int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
frame1.CopyFrom(frame2);
EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
EXPECT_EQ(frame2.muted(), frame1.muted());
EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
kNumChannels);
frame1.CopyFrom(frame2);
EXPECT_EQ(frame2.muted(), frame1.muted());
EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
}
} // namespace webrtc