/*
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* Copyright (c) 2013 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 "webrtc/modules/audio_device/fine_audio_buffer.h"
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#include <limits.h>
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#include <memory>
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
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using ::testing::_;
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using ::testing::AtLeast;
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using ::testing::InSequence;
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using ::testing::Return;
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namespace webrtc {
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// The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
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// to detect errors. This function verifies that the buffers contain such data.
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// E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
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// buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
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// will happen.
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// |buffer| is the audio buffer to verify.
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bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
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int start_value = (buffer_number * size) % SCHAR_MAX;
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for (int i = 0; i < size; ++i) {
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if (buffer[i] != (i + start_value) % SCHAR_MAX) {
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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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// This function replaces the real AudioDeviceBuffer::GetPlayoutData when it's
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// called (which is done implicitly when calling GetBufferData). It writes the
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// sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
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// buffer of different size than the one VerifyBuffer verifies.
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// |iteration| is the number of calls made to UpdateBuffer prior to this call.
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// |samples_per_10_ms| is the number of samples that should be written to the
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// buffer (|arg0|).
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ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
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int8_t* buffer = static_cast<int8_t*>(arg0);
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int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
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int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
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for (int i = 0; i < bytes_per_10_ms; ++i) {
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buffer[i] = (i + start_value) % SCHAR_MAX;
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}
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return samples_per_10_ms;
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}
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// Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
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// for details.
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void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
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int start_value = (iteration * size) % SCHAR_MAX;
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for (int i = 0; i < size; ++i) {
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buffer[i] = (i + start_value) % SCHAR_MAX;
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}
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}
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// Action macro which verifies that the recorded 10ms chunk of audio data
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// (in |arg0|) contains the correct reference values even if they have been
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// supplied using a buffer size that is smaller or larger than 10ms.
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// See VerifyBuffer() for details.
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ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
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const int8_t* buffer = static_cast<const int8_t*>(arg0);
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int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
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int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
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for (int i = 0; i < bytes_per_10_ms; ++i) {
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EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
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}
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return 0;
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}
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void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
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const int kSamplesPer10Ms = sample_rate * 10 / 1000;
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const int kFrameSizeBytes =
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frame_size_in_samples * static_cast<int>(sizeof(int16_t));
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const int kNumberOfFrames = 5;
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// Ceiling of integer division: 1 + ((x - 1) / y)
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const int kNumberOfUpdateBufferCalls =
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1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
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MockAudioDeviceBuffer audio_device_buffer;
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EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
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.WillRepeatedly(Return(kSamplesPer10Ms));
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{
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InSequence s;
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for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
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EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
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.WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
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.RetiresOnSaturation();
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}
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}
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{
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InSequence s;
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for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
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EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
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.WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
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.RetiresOnSaturation();
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}
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}
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EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
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.Times(kNumberOfUpdateBufferCalls - 1);
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EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
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.Times(kNumberOfUpdateBufferCalls - 1)
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.WillRepeatedly(Return(kSamplesPer10Ms));
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FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
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sample_rate);
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rtc::scoped_ptr<int8_t[]> out_buffer;
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out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
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rtc::scoped_ptr<int8_t[]> in_buffer;
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in_buffer.reset(new int8_t[kFrameSizeBytes]);
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for (int i = 0; i < kNumberOfFrames; ++i) {
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fine_buffer.GetPlayoutData(out_buffer.get());
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EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
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UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
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fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
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}
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}
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TEST(FineBufferTest, BufferLessThan10ms) {
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const int kSampleRate = 44100;
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const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
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const int kFrameSizeSamples = kSamplesPer10Ms - 50;
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RunFineBufferTest(kSampleRate, kFrameSizeSamples);
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}
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TEST(FineBufferTest, GreaterThan10ms) {
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const int kSampleRate = 44100;
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const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
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const int kFrameSizeSamples = kSamplesPer10Ms + 50;
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RunFineBufferTest(kSampleRate, kFrameSizeSamples);
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}
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} // namespace webrtc
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