tor-browser

fine_audio_buffer_unittest.cc (5788B)

---

/*
*  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 "modules/audio_device/fine_audio_buffer.h"

#include <climits>
#include <cstdint>
#include <memory>

#include "api/array_view.h"
#include "modules/audio_device/mock_audio_device_buffer.h"
#include "test/create_test_environment.h"
#include "test/gmock.h"
#include "test/gtest.h"

using ::testing::_;
using ::testing::AtLeast;
using ::testing::InSequence;
using ::testing::Return;

namespace webrtc {

constexpr int kSampleRate = 44100;
constexpr int kChannels = 2;
constexpr int kSamplesPer10Ms = kSampleRate * 10 / 1000;

// 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 int16_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 the real AudioDeviceBuffer::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) {
 int16_t* buffer = static_cast<int16_t*>(arg0);
 int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
 for (int i = 0; i < samples_per_10_ms; ++i) {
   buffer[i] = (i + start_value) % SCHAR_MAX;
 }
 // Should return samples per channel.
 return samples_per_10_ms / kChannels;
}

// Writes a periodic ramp pattern to the supplied `buffer`. See UpdateBuffer()
// for details.
void UpdateInputBuffer(int16_t* buffer, int iteration, int size) {
 int start_value = (iteration * size) % SCHAR_MAX;
 for (int i = 0; i < size; ++i) {
   buffer[i] = (i + start_value) % SCHAR_MAX;
 }
}

// Action macro which verifies that the recorded 10ms chunk of audio data
// (in `arg0`) contains the correct reference values even if they have been
// supplied using a buffer size that is smaller or larger than 10ms.
// See VerifyBuffer() for details.
ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
 const int16_t* buffer = static_cast<const int16_t*>(arg0);
 int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
 for (int i = 0; i < samples_per_10_ms; ++i) {
   EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
 }
 return 0;
}

void RunFineBufferTest(int frame_size_in_samples) {
 const int kFrameSizeSamples = frame_size_in_samples;
 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(CreateTestEnvironment());
 audio_device_buffer.SetPlayoutSampleRate(kSampleRate);
 audio_device_buffer.SetPlayoutChannels(kChannels);
 audio_device_buffer.SetRecordingSampleRate(kSampleRate);
 audio_device_buffer.SetRecordingChannels(kChannels);

 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, kChannels * kSamplesPer10Ms))
         .RetiresOnSaturation();
   }
 }
 {
   InSequence s;
   for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
     EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms, _))
         .WillOnce(VerifyInputBuffer(j, kChannels * kSamplesPer10Ms))
         .RetiresOnSaturation();
   }
 }
 EXPECT_CALL(audio_device_buffer, SetVQEData(_, _))
     .Times(kNumberOfUpdateBufferCalls - 1);
 EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
     .Times(kNumberOfUpdateBufferCalls - 1)
     .WillRepeatedly(Return(0));

 FineAudioBuffer fine_buffer(&audio_device_buffer);
 std::unique_ptr<int16_t[]> out_buffer(
     new int16_t[kChannels * kFrameSizeSamples]);
 std::unique_ptr<int16_t[]> in_buffer(
     new int16_t[kChannels * kFrameSizeSamples]);

 for (int i = 0; i < kNumberOfFrames; ++i) {
   fine_buffer.GetPlayoutData(
       ArrayView<int16_t>(out_buffer.get(), kChannels * kFrameSizeSamples), 0);
   EXPECT_TRUE(
       VerifyBuffer(out_buffer.get(), i, kChannels * kFrameSizeSamples));
   UpdateInputBuffer(in_buffer.get(), i, kChannels * kFrameSizeSamples);
   fine_buffer.DeliverRecordedData(
       ArrayView<const int16_t>(in_buffer.get(),
                                kChannels * kFrameSizeSamples),
       0);
 }
}

TEST(FineBufferTest, BufferLessThan10ms) {
 const int kFrameSizeSamples = kSamplesPer10Ms - 50;
 RunFineBufferTest(kFrameSizeSamples);
}

TEST(FineBufferTest, GreaterThan10ms) {
 const int kFrameSizeSamples = kSamplesPer10Ms + 50;
 RunFineBufferTest(kFrameSizeSamples);
}

}  // namespace webrtc