Index: content/renderer/media/speech_recognition_audio_sink_unittest.cc |
diff --git a/content/renderer/media/speech_recognition_audio_sink_unittest.cc b/content/renderer/media/speech_recognition_audio_sink_unittest.cc |
index 8a05fabd33acb87351fa3365b1b10b9f10236dbf..3ea81a8dea1fccfe5d15c9ed0dfd06775a992d85 100644 |
--- a/content/renderer/media/speech_recognition_audio_sink_unittest.cc |
+++ b/content/renderer/media/speech_recognition_audio_sink_unittest.cc |
@@ -4,7 +4,11 @@ |
#include "content/renderer/media/speech_recognition_audio_sink.h" |
+#include <stddef.h> |
+#include <stdint.h> |
+ |
#include "base/bind.h" |
+#include "base/macros.h" |
#include "base/strings/utf_string_conversions.h" |
#include "content/renderer/media/media_stream_audio_source.h" |
#include "content/renderer/media/mock_media_constraint_factory.h" |
@@ -49,7 +53,7 @@ class MockSyncSocket : public base::SyncSocket { |
struct SharedBuffer { |
SharedBuffer() : data(), start(0), length(0) {} |
- uint8 data[kSharedBufferSize]; |
+ uint8_t data[kSharedBufferSize]; |
size_t start; |
size_t length; |
}; |
@@ -87,7 +91,7 @@ size_t MockSyncSocket::Send(const void* buffer, size_t length) { |
if (in_failure_mode_) |
return 0; |
- const uint8* b = static_cast<const uint8*>(buffer); |
+ const uint8_t* b = static_cast<const uint8_t*>(buffer); |
for (size_t i = 0; i < length; ++i, ++buffer_->length) |
buffer_->data[buffer_->start + buffer_->length] = b[i]; |
@@ -96,7 +100,7 @@ size_t MockSyncSocket::Send(const void* buffer, size_t length) { |
} |
size_t MockSyncSocket::Receive(void* buffer, size_t length) { |
- uint8* b = static_cast<uint8*>(buffer); |
+ uint8_t* b = static_cast<uint8_t*>(buffer); |
for (size_t i = buffer_->start; i < buffer_->length; ++i, ++buffer_->start) |
b[i] = buffer_->data[buffer_->start]; |
@@ -117,7 +121,7 @@ class FakeSpeechRecognizer { |
const media::AudioParameters& sink_params, |
base::SharedMemoryHandle* foreign_memory_handle) { |
// Shared memory is allocated, mapped and shared. |
- const uint32 kSharedMemorySize = |
+ const uint32_t kSharedMemorySize = |
sizeof(media::AudioInputBufferParameters) + |
media::AudioBus::CalculateMemorySize(sink_params); |
shared_memory_.reset(new base::SharedMemory()); |
@@ -211,10 +215,10 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
// Initializes the producer and consumer with specified audio parameters. |
// Returns the minimal number of input audio buffers which need to be captured |
// before they get sent to the consumer. |
- uint32 Initialize(int input_sample_rate, |
- int input_frames_per_buffer, |
- int output_sample_rate, |
- int output_frames_per_buffer) { |
+ uint32_t Initialize(int input_sample_rate, |
+ int input_frames_per_buffer, |
+ int output_sample_rate, |
+ int output_frames_per_buffer) { |
// Audio Environment setup. |
source_params_.Reset(kInputFormat, |
kInputChannelLayout, |
@@ -255,7 +259,7 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
base::Unretained(this)))); |
// Return number of buffers needed to trigger resampling and consumption. |
- return static_cast<uint32>(std::ceil( |
+ return static_cast<uint32_t>(std::ceil( |
static_cast<double>(output_frames_per_buffer * input_sample_rate) / |
(input_frames_per_buffer * output_sample_rate))); |
} |
@@ -295,8 +299,8 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
} |
// Emulates an audio capture device capturing data from the source. |
- inline void CaptureAudio(const uint32 buffers) { |
- for (uint32 i = 0; i < buffers; ++i) { |
+ inline void CaptureAudio(const uint32_t buffers) { |
+ for (uint32_t i = 0; i < buffers; ++i) { |
const base::TimeTicks estimated_capture_time = first_frame_capture_time_ + |
(sample_frames_captured_ * base::TimeDelta::FromSeconds(1) / |
source_params_.sample_rate()); |
@@ -311,14 +315,15 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
} |
// Helper method for verifying captured audio data has been consumed. |
- inline void AssertConsumedBuffers(const uint32 buffer_index) { |
+ inline void AssertConsumedBuffers(const uint32_t buffer_index) { |
ASSERT_EQ(buffer_index, recognizer()->GetAudioInputBuffer()->params.size); |
} |
// Helper method for providing audio data to producer and verifying it was |
// consumed on the recognizer. |
- inline void CaptureAudioAndAssertConsumedBuffers(const uint32 buffers, |
- const uint32 buffer_index) { |
+ inline void CaptureAudioAndAssertConsumedBuffers( |
+ const uint32_t buffers, |
+ const uint32_t buffer_index) { |
CaptureAudio(buffers); |
AssertConsumedBuffers(buffer_index); |
} |
@@ -330,15 +335,13 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
const int input_frames_per_buffer, |
const int output_sample_rate, |
const int output_frames_per_buffer, |
- const uint32 consumptions) { |
- const uint32 buffers_per_notification = |
- Initialize(input_sample_rate, |
- input_frames_per_buffer, |
- output_sample_rate, |
- output_frames_per_buffer); |
+ const uint32_t consumptions) { |
+ const uint32_t buffers_per_notification = |
+ Initialize(input_sample_rate, input_frames_per_buffer, |
+ output_sample_rate, output_frames_per_buffer); |
AssertConsumedBuffers(0U); |
- for (uint32 i = 1U; i <= consumptions; ++i) { |
+ for (uint32_t i = 1U; i <= consumptions; ++i) { |
CaptureAudio(buffers_per_notification); |
ASSERT_EQ(i, recognizer()->GetAudioInputBuffer()->params.size) |
<< "Tested at rates: " |
@@ -371,7 +374,7 @@ class SpeechRecognitionAudioSinkTest : public testing::Test { |
WebRtcLocalAudioTrack* native_track_; |
base::TimeTicks first_frame_capture_time_; |
- int64 sample_frames_captured_; |
+ int64_t sample_frames_captured_; |
DISALLOW_COPY_AND_ASSIGN(SpeechRecognitionAudioSinkTest); |
}; |
@@ -433,12 +436,12 @@ TEST_F(SpeechRecognitionAudioSinkTest, AudioDataIsResampledOnSink) { |
// Input audio is sampled at 44.1 KHz with data chunks of 10ms. Desired output |
// is corresponding to the speech recognition engine requirements: 16 KHz with |
// 100 ms chunks (1600 frames per buffer). |
- const uint32 kSourceFrames = 441; |
- const uint32 buffers_per_notification = |
+ const uint32_t kSourceFrames = 441; |
+ const uint32_t buffers_per_notification = |
Initialize(44100, kSourceFrames, 16000, 1600); |
// Fill audio input frames with 0, 1, 2, 3, ..., 440. |
- int16 source_data[kSourceFrames * kInputChannels]; |
- for (uint32 i = 0; i < kSourceFrames; ++i) { |
+ int16_t source_data[kSourceFrames * kInputChannels]; |
+ for (uint32_t i = 0; i < kSourceFrames; ++i) { |
for (int c = 0; c < kInputChannels; ++c) |
source_data[i * kInputChannels + c] = i; |
} |
@@ -447,18 +450,18 @@ TEST_F(SpeechRecognitionAudioSinkTest, AudioDataIsResampledOnSink) { |
// Prepare sink audio bus and data for rendering. |
media::AudioBus* sink_bus = recognizer()->audio_bus(); |
- const uint32 kSinkDataLength = 1600 * kOutputChannels; |
- int16 sink_data[kSinkDataLength] = {0}; |
+ const uint32_t kSinkDataLength = 1600 * kOutputChannels; |
+ int16_t sink_data[kSinkDataLength] = {0}; |
// Render the audio data from the recognizer. |
sink_bus->ToInterleaved(sink_bus->frames(), |
sink_params().bits_per_sample() / 8, sink_data); |
// Checking only a fraction of the sink frames. |
- const uint32 kNumFramesToTest = 12; |
+ const uint32_t kNumFramesToTest = 12; |
// Check all channels are zeroed out before we trigger resampling. |
- for (uint32 i = 0; i < kNumFramesToTest; ++i) { |
+ for (uint32_t i = 0; i < kNumFramesToTest; ++i) { |
for (int c = 0; c < kOutputChannels; ++c) |
EXPECT_EQ(0, sink_data[i * kOutputChannels + c]); |
} |
@@ -472,11 +475,11 @@ TEST_F(SpeechRecognitionAudioSinkTest, AudioDataIsResampledOnSink) { |
sink_params().bits_per_sample() / 8, sink_data); |
// Resampled data expected frames. Extracted based on |source_data|. |
- const int16 kExpectedData[kNumFramesToTest] = {0, 2, 5, 8, 11, 13, |
- 16, 19, 22, 24, 27, 30}; |
+ const int16_t kExpectedData[kNumFramesToTest] = {0, 2, 5, 8, 11, 13, |
+ 16, 19, 22, 24, 27, 30}; |
// Check all channels have the same resampled data. |
- for (uint32 i = 0; i < kNumFramesToTest; ++i) { |
+ for (uint32_t i = 0; i < kNumFramesToTest; ++i) { |
for (int c = 0; c < kOutputChannels; ++c) |
EXPECT_EQ(kExpectedData[i], sink_data[i * kOutputChannels + c]); |
} |
@@ -484,7 +487,7 @@ TEST_F(SpeechRecognitionAudioSinkTest, AudioDataIsResampledOnSink) { |
// Checks that the producer does not misbehave when a socket failure occurs. |
TEST_F(SpeechRecognitionAudioSinkTest, SyncSocketFailsSendingData) { |
- const uint32 buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
+ const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
// Start with no problems on the socket. |
AssertConsumedBuffers(0U); |
CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); |
@@ -499,7 +502,7 @@ TEST_F(SpeechRecognitionAudioSinkTest, SyncSocketFailsSendingData) { |
// We check that the FIFO overflow does not occur and that the producer is able |
// to resume. |
TEST_F(SpeechRecognitionAudioSinkTest, RepeatedSycnhronizationLag) { |
- const uint32 buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
+ const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
// Start with no synchronization problems. |
AssertConsumedBuffers(0U); |
@@ -520,7 +523,7 @@ TEST_F(SpeechRecognitionAudioSinkTest, RepeatedSycnhronizationLag) { |
// Checks that an OnStoppedCallback is issued when the track is stopped. |
TEST_F(SpeechRecognitionAudioSinkTest, OnReadyStateChangedOccured) { |
- const uint32 buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
+ const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
AssertConsumedBuffers(0U); |
CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); |
EXPECT_CALL(*this, StoppedCallback()).Times(1); |