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/*
* Copyright (c) 2013 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 "webrtc/modules/audio_device/android/fine_audio_buffer.h"
#include <limits.h>
#include <memory>
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using ::testing::_;
using ::testing::InSequence;
using ::testing::Return;
namespace webrtc {
// The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
// to detect errors. This function verifies that the buffers contain such data.
// E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
// buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
// will happen.
// |buffer| is the audio buffer to verify.
bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
int start_value = (buffer_number * size) % SCHAR_MAX;
for (int i = 0; i < size; ++i) {
if (buffer[i] != (i + start_value) % SCHAR_MAX) {
return false;
}
}
return true;
}
// This function replaces GetPlayoutData when it's called (which is done
// implicitly when calling GetBufferData). It writes the sequence
// 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a buffer of
// different size than the one VerifyBuffer verifies.
// |iteration| is the number of calls made to UpdateBuffer prior to this call.
// |samples_per_10_ms| is the number of samples that should be written to the
// buffer (|arg0|).
ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
int8_t* buffer = static_cast<int8_t*>(arg0);
int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
for (int i = 0; i < bytes_per_10_ms; ++i) {
buffer[i] = (i + start_value) % SCHAR_MAX;
}
return samples_per_10_ms;
}
void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
const int kSamplesPer10Ms = sample_rate * 10 / 1000;
const int kFrameSizeBytes = frame_size_in_samples *
static_cast<int>(sizeof(int16_t));
const int kNumberOfFrames = 5;
// Ceiling of integer division: 1 + ((x - 1) / y)
const int kNumberOfUpdateBufferCalls =
1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
MockAudioDeviceBuffer audio_device_buffer;
EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
.WillRepeatedly(Return(kSamplesPer10Ms));
{
InSequence s;
for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
.WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
.RetiresOnSaturation();
}
}
FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
sample_rate);
scoped_ptr<int8_t[]> out_buffer;
out_buffer.reset(
new int8_t[fine_buffer.RequiredBufferSizeBytes()]);
for (int i = 0; i < kNumberOfFrames; ++i) {
fine_buffer.GetBufferData(out_buffer.get());
EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
}
}
TEST(FineBufferTest, BufferLessThan10ms) {
const int kSampleRate = 44100;
const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
const int kFrameSizeSamples = kSamplesPer10Ms - 50;
RunFineBufferTest(kSampleRate, kFrameSizeSamples);
}
TEST(FineBufferTest, GreaterThan10ms) {
const int kSampleRate = 44100;
const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
const int kFrameSizeSamples = kSamplesPer10Ms + 50;
RunFineBufferTest(kSampleRate, kFrameSizeSamples);
}
} // namespace webrtc
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