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| 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #include "content/renderer/media/speech_recognition_audio_sink.h" |
| 6 |
| 7 #include "base/strings/utf_string_conversions.h" |
| 8 #include "content/renderer/media/mock_media_constraint_factory.h" |
| 9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| 10 #include "content/renderer/media/webrtc_local_audio_track.h" |
| 11 #include "media/audio/audio_parameters.h" |
| 12 #include "media/base/audio_bus.h" |
| 13 #include "testing/gmock/include/gmock/gmock.h" |
| 14 #include "testing/gtest/include/gtest/gtest.h" |
| 15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| 16 |
| 17 namespace { |
| 18 |
| 19 // Supported speech recognition audio parameters. |
| 20 const int kSpeechRecognitionSampleRate = 16000; |
| 21 const int kSpeechRecognitionFramesPerBuffer = 1600; |
| 22 |
| 23 // Input audio format. |
| 24 const media::AudioParameters::Format kInputFormat = |
| 25 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; |
| 26 const media::ChannelLayout kInputChannelLayout = media::CHANNEL_LAYOUT_MONO; |
| 27 const int kInputChannels = 1; |
| 28 const int kInputBitsPerSample = 16; |
| 29 |
| 30 // Output audio format. |
| 31 const media::AudioParameters::Format kOutputFormat = |
| 32 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; |
| 33 const media::ChannelLayout kOutputChannelLayout = media::CHANNEL_LAYOUT_STEREO; |
| 34 const int kOutputChannels = 2; |
| 35 const int kOutputBitsPerSample = 16; |
| 36 |
| 37 // Mocked out sockets used for Send/Receive. |
| 38 // Data is written and read from a shared buffer used as a FIFO and there is |
| 39 // no blocking. |OnSendCB| is used to trigger a |Receive| on the other socket. |
| 40 class MockSyncSocket : public base::SyncSocket { |
| 41 public: |
| 42 // This allows for 2 requests in queue between the |MockSyncSocket|s. |
| 43 static const int kSharedBufferSize = 8; |
| 44 |
| 45 // Buffer to be shared between two |MockSyncSocket|s. Allocated on heap. |
| 46 struct SharedBuffer { |
| 47 SharedBuffer() : data(), start(0), length(0) {} |
| 48 |
| 49 uint8 data[kSharedBufferSize]; |
| 50 size_t start; |
| 51 size_t length; |
| 52 }; |
| 53 |
| 54 // Callback used for pairing an A.Send() with B.Receieve() without blocking. |
| 55 typedef base::Callback<void()> OnSendCB; |
| 56 |
| 57 explicit MockSyncSocket(SharedBuffer* shared_buffer) |
| 58 : buffer_(shared_buffer), |
| 59 in_failure_mode_(false) {} |
| 60 |
| 61 MockSyncSocket(SharedBuffer* shared_buffer, const OnSendCB& on_send_cb) |
| 62 : buffer_(shared_buffer), |
| 63 on_send_cb_(on_send_cb), |
| 64 in_failure_mode_(false) {} |
| 65 |
| 66 virtual size_t Send(const void* buffer, size_t length) OVERRIDE; |
| 67 virtual size_t Receive(void* buffer, size_t length) OVERRIDE; |
| 68 |
| 69 // When |in_failure_mode_| == true, the socket fails to send. |
| 70 void SetFailureMode(bool in_failure_mode) { |
| 71 in_failure_mode_ = in_failure_mode; |
| 72 } |
| 73 |
| 74 private: |
| 75 SharedBuffer* buffer_; |
| 76 const OnSendCB on_send_cb_; |
| 77 bool in_failure_mode_; |
| 78 }; |
| 79 |
| 80 size_t MockSyncSocket::Send(const void* buffer, size_t length) { |
| 81 if (in_failure_mode_) |
| 82 return 0; |
| 83 |
| 84 const uint8* b = static_cast<const uint8*>(buffer); |
| 85 for (size_t i = 0; i < length; ++i, ++buffer_->length) |
| 86 buffer_->data[buffer_->start + buffer_->length] = b[i]; |
| 87 |
| 88 on_send_cb_.Run(); |
| 89 return length; |
| 90 } |
| 91 |
| 92 size_t MockSyncSocket::Receive(void* buffer, size_t length) { |
| 93 uint8* b = static_cast<uint8*>(buffer); |
| 94 for (size_t i = buffer_->start; i < buffer_->length; ++i, ++buffer_->start) |
| 95 b[i] = buffer_->data[buffer_->start]; |
| 96 |
| 97 // Since buffer is used sequentially, we can reset the buffer indices here. |
| 98 buffer_->start = buffer_->length = 0; |
| 99 return length; |
| 100 } |
| 101 |
| 102 // This fake class is the consumer used to verify behaviour of the producer. |
| 103 // The |Initialize()| method shows what the consumer should be responsible for |
| 104 // in the production code (minus the mocks). |
| 105 class FakeSpeechRecognizer { |
| 106 public: |
| 107 FakeSpeechRecognizer() : is_responsive_(true) { } |
| 108 |
| 109 void Initialize( |
| 110 const blink::WebMediaStreamTrack& track, |
| 111 const media::AudioParameters& sink_params, |
| 112 base::SharedMemoryHandle* foreign_memory_handle) { |
| 113 // Shared memory is allocated, mapped and shared. |
| 114 uint32 shared_memory_size = |
| 115 sizeof(media::AudioInputBufferParameters) + |
| 116 media::AudioBus::CalculateMemorySize(sink_params); |
| 117 shared_memory_.reset(new base::SharedMemory()); |
| 118 ASSERT_TRUE(shared_memory_->CreateAndMapAnonymous(shared_memory_size)); |
| 119 ASSERT_TRUE(shared_memory_->ShareToProcess(base::GetCurrentProcessHandle(), |
| 120 foreign_memory_handle)); |
| 121 |
| 122 // Wrap the shared memory for the audio bus. |
| 123 media::AudioInputBuffer* buffer = |
| 124 static_cast<media::AudioInputBuffer*>(shared_memory_->memory()); |
| 125 audio_track_bus_ = media::AudioBus::WrapMemory(sink_params, buffer->audio); |
| 126 |
| 127 // Reference to the counter used to synchronize. |
| 128 buffer_index_ = &(buffer->params.size); |
| 129 *buffer_index_ = 0U; |
| 130 |
| 131 // Create a shared buffer for the |MockSyncSocket|s. |
| 132 shared_buffer_.reset(new MockSyncSocket::SharedBuffer()); |
| 133 |
| 134 // Local socket will receive signals from the producer. |
| 135 local_socket_.reset(new MockSyncSocket(shared_buffer_.get())); |
| 136 |
| 137 // We automatically trigger a Receive when data is sent over the socket. |
| 138 foreign_socket_ = new MockSyncSocket( |
| 139 shared_buffer_.get(), |
| 140 base::Bind(&FakeSpeechRecognizer::EmulateReceiveThreadLoopIteration, |
| 141 base::Unretained(this))); |
| 142 |
| 143 // This is usually done to pair the sockets. Here it's not effective. |
| 144 base::SyncSocket::CreatePair(local_socket_.get(), foreign_socket_); |
| 145 } |
| 146 |
| 147 // Emulates a single iteraton of a thread receiving on the socket. |
| 148 // This would normally be done on a receiving thread's task on the browser. |
| 149 void EmulateReceiveThreadLoopIteration() { |
| 150 // When not responsive do nothing as if the process is busy. |
| 151 if (!is_responsive_) |
| 152 return; |
| 153 |
| 154 local_socket_->Receive(buffer_index_, sizeof(*buffer_index_)); |
| 155 // Notify the producer that the audio buffer has been consumed. |
| 156 ++(*buffer_index_); |
| 157 } |
| 158 |
| 159 // Used to simulate an unresponsive behaviour of the consumer. |
| 160 void SimulateResponsiveness(bool is_responsive) { |
| 161 is_responsive_ = is_responsive; |
| 162 } |
| 163 |
| 164 MockSyncSocket* foreign_socket() { return foreign_socket_; } |
| 165 media::AudioBus* audio_bus() const { return audio_track_bus_.get(); } |
| 166 uint32 buffer_index() { return *buffer_index_; } |
| 167 |
| 168 private: |
| 169 bool is_responsive_; |
| 170 |
| 171 // Shared memory for the audio and synchronization. |
| 172 scoped_ptr<base::SharedMemory> shared_memory_; |
| 173 |
| 174 // Fake sockets and their shared buffer. |
| 175 scoped_ptr<MockSyncSocket::SharedBuffer> shared_buffer_; |
| 176 scoped_ptr<MockSyncSocket> local_socket_; |
| 177 MockSyncSocket* foreign_socket_; |
| 178 |
| 179 // Audio bus wrapping the shared memory from the renderer. |
| 180 scoped_ptr<media::AudioBus> audio_track_bus_; |
| 181 |
| 182 // Used for synchronization of sent/received buffers. |
| 183 uint32* buffer_index_; |
| 184 }; |
| 185 |
| 186 } // namespace |
| 187 |
| 188 namespace content { |
| 189 |
| 190 class SpeechRecognitionAudioSinkTest : public testing::Test { |
| 191 public: |
| 192 SpeechRecognitionAudioSinkTest() { } |
| 193 |
| 194 // Initializes the producer and consumer with specified audio parameters. |
| 195 // Returns the minimal number of input audio buffers which need to be captured |
| 196 // before they get sent to the consumer. |
| 197 uint32 Initialize(int input_sample_rate, |
| 198 int input_frames_per_buffer, |
| 199 int output_sample_rate, |
| 200 int output_frames_per_buffer) { |
| 201 // Audio Environment setup. |
| 202 source_params_.Reset(kInputFormat, |
| 203 kInputChannelLayout, |
| 204 kInputChannels, |
| 205 input_sample_rate, |
| 206 kInputBitsPerSample, |
| 207 input_frames_per_buffer); |
| 208 sink_params_.Reset(kOutputFormat, |
| 209 kOutputChannelLayout, |
| 210 kOutputChannels, |
| 211 output_sample_rate, |
| 212 kOutputBitsPerSample, |
| 213 output_frames_per_buffer); |
| 214 source_data_.reset(new int16[input_frames_per_buffer * kInputChannels]); |
| 215 |
| 216 // Prepare the track and audio source. |
| 217 blink::WebMediaStreamTrack blink_track; |
| 218 PrepareBlinkTrackOfType(MEDIA_DEVICE_AUDIO_CAPTURE, &blink_track); |
| 219 |
| 220 // Get the native track from the blink track and initialize. |
| 221 native_track_ = |
| 222 static_cast<WebRtcLocalAudioTrack*>(blink_track.extraData()); |
| 223 native_track_->OnSetFormat(source_params_); |
| 224 |
| 225 // Create and initialize the consumer. |
| 226 recognizer_.reset(new FakeSpeechRecognizer()); |
| 227 base::SharedMemoryHandle foreign_memory_handle; |
| 228 recognizer_->Initialize(blink_track, sink_params_, &foreign_memory_handle); |
| 229 |
| 230 // Create the producer. |
| 231 scoped_ptr<base::SyncSocket> foreign_socket(recognizer_->foreign_socket()); |
| 232 speech_audio_sink_.reset(new SpeechRecognitionAudioSink( |
| 233 blink_track, sink_params_, foreign_memory_handle, |
| 234 foreign_socket.Pass(), |
| 235 base::Bind(&SpeechRecognitionAudioSinkTest::StoppedCallback, |
| 236 base::Unretained(this)))); |
| 237 |
| 238 // Return number of buffers needed to trigger resampling and consumption. |
| 239 return static_cast<uint32>(std::ceil( |
| 240 static_cast<double>(output_frames_per_buffer * input_sample_rate) / |
| 241 (input_frames_per_buffer * output_sample_rate))); |
| 242 } |
| 243 |
| 244 // Mock callback expected to be called when the track is stopped. |
| 245 MOCK_METHOD0(StoppedCallback, void()); |
| 246 |
| 247 protected: |
| 248 // Prepares a blink track of a given MediaStreamType and attaches the native |
| 249 // track which can be used to capture audio data and pass it to the producer. |
| 250 static void PrepareBlinkTrackOfType( |
| 251 const MediaStreamType device_type, |
| 252 blink::WebMediaStreamTrack* blink_track) { |
| 253 StreamDeviceInfo device_info(device_type, "Mock device", |
| 254 "mock_device_id"); |
| 255 MockMediaConstraintFactory constraint_factory; |
| 256 const blink::WebMediaConstraints constraints = |
| 257 constraint_factory.CreateWebMediaConstraints(); |
| 258 scoped_refptr<WebRtcAudioCapturer> capturer( |
| 259 WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL, |
| 260 NULL)); |
| 261 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| 262 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| 263 scoped_ptr<WebRtcLocalAudioTrack> native_track( |
| 264 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
| 265 blink::WebMediaStreamSource blink_audio_source; |
| 266 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), |
| 267 blink::WebMediaStreamSource::TypeAudio, |
| 268 base::UTF8ToUTF16("dummy_source_name")); |
| 269 MediaStreamSource::SourceStoppedCallback cb; |
| 270 blink_audio_source.setExtraData( |
| 271 new MediaStreamAudioSource(-1, device_info, cb, NULL)); |
| 272 blink_track->initialize(blink::WebString::fromUTF8("dummy_track"), |
| 273 blink_audio_source); |
| 274 blink_track->setExtraData(native_track.release()); |
| 275 } |
| 276 |
| 277 // Emulates an audio capture device capturing data from the source. |
| 278 inline void CaptureAudio(const uint32 buffers) { |
| 279 for (uint32 i = 0; i < buffers; ++i) |
| 280 native_track_->Capture(source_data_.get(), |
| 281 base::TimeDelta::FromMilliseconds(0), 1, false, |
| 282 false); |
| 283 } |
| 284 |
| 285 // Used to simulate a problem with sockets. |
| 286 void SetFailureModeOnForeignSocket(bool in_failure_mode) { |
| 287 recognizer_->foreign_socket()->SetFailureMode(in_failure_mode); |
| 288 } |
| 289 |
| 290 // Helper method for verifying captured audio data has been consumed. |
| 291 inline void AssertConsumedBuffers(const uint32 buffer_index) { |
| 292 ASSERT_EQ(buffer_index, recognizer_->buffer_index()); |
| 293 } |
| 294 |
| 295 // Helper method for providing audio data to producer and verifying it was |
| 296 // consumed on the recognizer. |
| 297 inline void CaptureAudioAndAssertConsumedBuffers(const uint32 buffers, |
| 298 const uint32 buffer_index) { |
| 299 CaptureAudio(buffers); |
| 300 AssertConsumedBuffers(buffer_index); |
| 301 } |
| 302 |
| 303 // Helper method to capture and assert consumption at different sample rates |
| 304 // and audio buffer sizes. |
| 305 inline void AssertConsumptionForAudioParameters( |
| 306 const int input_sample_rate, |
| 307 const int input_frames_per_buffer, |
| 308 const int output_sample_rate, |
| 309 const int output_frames_per_buffer, |
| 310 const uint32 consumptions) { |
| 311 const uint32 kBuffersPerNotification = |
| 312 Initialize(input_sample_rate, input_frames_per_buffer, |
| 313 output_sample_rate, output_frames_per_buffer); |
| 314 AssertConsumedBuffers(0U); |
| 315 |
| 316 for (uint32 i = 1U; i <= consumptions; ++i) { |
| 317 CaptureAudio(kBuffersPerNotification); |
| 318 ASSERT_EQ(i, recognizer_->buffer_index()) |
| 319 << "Tested at rates: " |
| 320 << "In(" << input_sample_rate << ", " << input_frames_per_buffer |
| 321 << ") " |
| 322 << "Out(" << output_sample_rate << ", " << output_frames_per_buffer |
| 323 << ")"; |
| 324 } |
| 325 } |
| 326 |
| 327 // Producer. |
| 328 scoped_ptr<SpeechRecognitionAudioSink> speech_audio_sink_; |
| 329 |
| 330 // Consumer. |
| 331 scoped_ptr<FakeSpeechRecognizer> recognizer_; |
| 332 |
| 333 // Audio related members. |
| 334 scoped_ptr<int16[]> source_data_; |
| 335 media::AudioParameters source_params_; |
| 336 media::AudioParameters sink_params_; |
| 337 WebRtcLocalAudioTrack* native_track_; |
| 338 }; |
| 339 |
| 340 // Not all types of tracks are supported. This test checks if that policy is |
| 341 // implemented correctly. |
| 342 TEST_F(SpeechRecognitionAudioSinkTest, CheckIsSupportedAudioTrack) { |
| 343 typedef std::map<MediaStreamType, bool> SupportedTrackPolicy; |
| 344 |
| 345 // This test must be aligned with the policy of supported tracks. |
| 346 SupportedTrackPolicy p; |
| 347 p[MEDIA_NO_SERVICE] = false; |
| 348 p[MEDIA_DEVICE_AUDIO_CAPTURE] = true; // The only one supported for now. |
| 349 p[MEDIA_DEVICE_VIDEO_CAPTURE] = false; |
| 350 p[MEDIA_TAB_AUDIO_CAPTURE] = false; |
| 351 p[MEDIA_TAB_VIDEO_CAPTURE] = false; |
| 352 p[MEDIA_DESKTOP_VIDEO_CAPTURE] = false; |
| 353 p[MEDIA_LOOPBACK_AUDIO_CAPTURE] = false; |
| 354 p[MEDIA_DEVICE_AUDIO_OUTPUT] = false; |
| 355 |
| 356 // Ensure this test gets updated along with |content::MediaStreamType| enum. |
| 357 EXPECT_EQ(NUM_MEDIA_TYPES, p.size()); |
| 358 |
| 359 // Check the the entire policy. |
| 360 for (SupportedTrackPolicy::iterator it = p.begin(); it != p.end(); ++it) { |
| 361 blink::WebMediaStreamTrack blink_track; |
| 362 PrepareBlinkTrackOfType(it->first, &blink_track); |
| 363 ASSERT_EQ( |
| 364 it->second, |
| 365 SpeechRecognitionAudioSink::IsSupportedTrack(blink_track)); |
| 366 } |
| 367 } |
| 368 |
| 369 // Checks if the producer can support the listed range of input sample rates |
| 370 // and associated buffer sizes. |
| 371 TEST_F(SpeechRecognitionAudioSinkTest, RecognizerNotifiedOnSocket) { |
| 372 const size_t kNumAudioParamTuples = 24; |
| 373 const int kAudioParams[kNumAudioParamTuples][2] = { |
| 374 {8000, 80}, {8000, 800}, {16000, 160}, {16000, 1600}, |
| 375 {24000, 240}, {24000, 2400}, {32000, 320}, {32000, 3200}, |
| 376 {44100, 441}, {44100, 4410}, {48000, 480}, {48000, 4800}, |
| 377 {96000, 960}, {96000, 9600}, {11025, 111}, {11025, 1103}, |
| 378 {22050, 221}, {22050, 2205}, {88200, 882}, {88200, 8820}, |
| 379 {176400, 1764}, {176400, 17640}, {192000, 1920}, {192000, 19200}}; |
| 380 |
| 381 // Check all listed tuples of input sample rates and buffers sizes. |
| 382 for (size_t i = 0; i < kNumAudioParamTuples; ++i) { |
| 383 AssertConsumptionForAudioParameters( |
| 384 kAudioParams[i][0], kAudioParams[i][1], |
| 385 kSpeechRecognitionSampleRate, kSpeechRecognitionFramesPerBuffer, 3U); |
| 386 } |
| 387 } |
| 388 |
| 389 // Checks that the input data is getting resampled to the target sample rate. |
| 390 TEST_F(SpeechRecognitionAudioSinkTest, AudioDataIsResampledOnSink) { |
| 391 EXPECT_GE(kInputChannels, 1); |
| 392 EXPECT_GE(kOutputChannels, 1); |
| 393 |
| 394 // Input audio is sampled at 44.1 KHz with data chunks of 10ms. Desired output |
| 395 // is corresponding to the speech recognition engine requirements: 16 KHz with |
| 396 // 100 ms chunks (1600 frames per buffer). |
| 397 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); |
| 398 |
| 399 // Fill audio input frames with 0, 1, 2, 3, ..., 440. |
| 400 const uint32 kSourceDataLength = 441 * kInputChannels; |
| 401 for (uint32 i = 0; i < kSourceDataLength; ++i) { |
| 402 for (int c = 0; c < kInputChannels; ++c) |
| 403 source_data_[i * kInputChannels + c] = i; |
| 404 } |
| 405 |
| 406 // Prepare sink audio bus and data for rendering. |
| 407 media::AudioBus* sink_bus = recognizer_->audio_bus(); |
| 408 const uint32 kSinkDataLength = 1600 * kOutputChannels; |
| 409 int16 sink_data[kSinkDataLength] = {0}; |
| 410 |
| 411 // Render the audio data from the recognizer. |
| 412 sink_bus->ToInterleaved(sink_bus->frames(), |
| 413 sink_params_.bits_per_sample() / 8, sink_data); |
| 414 |
| 415 // Checking only a fraction of the sink frames. |
| 416 const uint32 kNumFramesToTest = 12; |
| 417 |
| 418 // Check all channels are zeroed out before we trigger resampling. |
| 419 for (uint32 i = 0; i < kNumFramesToTest; ++i) { |
| 420 for (int c = 0; c < kOutputChannels; ++c) |
| 421 EXPECT_EQ(0, sink_data[i * kOutputChannels + c]); |
| 422 } |
| 423 |
| 424 // Trigger the speech sink to resample the input data. |
| 425 AssertConsumedBuffers(0U); |
| 426 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); |
| 427 |
| 428 // Render the audio data from the recognizer. |
| 429 sink_bus->ToInterleaved(sink_bus->frames(), |
| 430 sink_params_.bits_per_sample() / 8, sink_data); |
| 431 |
| 432 // Resampled data expected frames. Extracted based on |source_data_|. |
| 433 const int16 kExpectedData[kNumFramesToTest] = {0, 2, 5, 8, 11, 13, |
| 434 16, 19, 22, 24, 27, 30}; |
| 435 |
| 436 // Check all channels have the same resampled data. |
| 437 for (uint32 i = 0; i < kNumFramesToTest; ++i) { |
| 438 for (int c = 0; c < kOutputChannels; ++c) |
| 439 EXPECT_EQ(kExpectedData[i], sink_data[i * kOutputChannels + c]); |
| 440 } |
| 441 } |
| 442 |
| 443 // Checks that the producer does not misbehave when a socket failure occurs. |
| 444 TEST_F(SpeechRecognitionAudioSinkTest, SyncSocketFailsSendingData) { |
| 445 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); |
| 446 // Start with no problems on the socket. |
| 447 AssertConsumedBuffers(0U); |
| 448 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); |
| 449 |
| 450 // A failure occurs (socket cannot send). |
| 451 SetFailureModeOnForeignSocket(true); |
| 452 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); |
| 453 } |
| 454 |
| 455 // Checks that an OnStoppedCallback is issued when the track is stopped. |
| 456 TEST_F(SpeechRecognitionAudioSinkTest, OnReadyStateChangedOccured) { |
| 457 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); |
| 458 AssertConsumedBuffers(0U); |
| 459 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); |
| 460 EXPECT_CALL(*this, StoppedCallback()).Times(1); |
| 461 |
| 462 native_track_->Stop(); |
| 463 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); |
| 464 } |
| 465 |
| 466 } // namespace content |
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