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Unified Diff: content/renderer/media/speech_recognition_audio_sink_unittest.cc

Issue 1547073003: Switch to standard integer types in content/renderer/. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 5 years ago
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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);
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