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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/speech_recognition_audio_source_provider.h" | |
6 | |
7 #include "base/logging.h" | |
8 #include "base/strings/utf_string_conversions.h" | |
9 #include "content/renderer/media/media_stream_audio_source.h" | |
10 #include "content/renderer/media/mock_media_constraint_factory.h" | |
11 #include "content/renderer/media/rtc_media_constraints.h" | |
12 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory. h" | |
13 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
14 #include "content/renderer/media/webrtc_audio_capturer.h" | |
15 #include "content/renderer/media/webrtc_audio_device_impl.h" | |
16 #include "content/renderer/media/webrtc_local_audio_source_provider.h" | |
17 #include "content/renderer/media/webrtc_local_audio_track.h" | |
18 #include "media/audio/audio_parameters.h" | |
19 #include "media/base/audio_bus.h" | |
20 #include "media/base/audio_capturer_source.h" | |
21 #include "testing/gmock/include/gmock/gmock.h" | |
22 #include "testing/gtest/include/gtest/gtest.h" | |
23 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | |
24 | |
25 namespace content { | |
26 | |
27 //////////////////////////////////////////////////////////////////////////////// | |
28 | |
29 // Buffer to be shared between two fake sockets. | |
burnik
2014/09/12 12:09:12
'fake' is interchangeable with 'mock' regarding so
burnik
2014/09/15 15:00:07
Done.
| |
30 struct SharedBuffer { | |
31 uint8 data[100000]; | |
no longer working on chromium
2014/09/15 08:31:29
noooooo, you can't allocate 100000 bytes in stack
burnik
2014/09/15 15:00:07
This is allocated on and owned by the FakeSpeechRe
| |
32 size_t start; | |
33 size_t length; | |
34 }; | |
35 | |
36 //////////////////////////////////////////////////////////////////////////////// | |
37 | |
38 // Fake socket used for Send/Receive. | |
39 // Data is written and read from a shared buffer used as a FIFO and there is | |
40 // no blocking. |OnSendCB| is used to trigger a |Receive| on the other socket. | |
41 class MockSyncSocket : public base::SyncSocket { | |
42 public: | |
43 typedef base::Callback<void()> OnSendCB; | |
44 | |
45 explicit MockSyncSocket(SharedBuffer* shared_buffer); | |
46 MockSyncSocket(SharedBuffer* shared_buffer, const OnSendCB& on_send_cb); | |
47 | |
48 virtual size_t Send(const void* buffer, size_t length) OVERRIDE; | |
49 virtual size_t Receive(void* buffer, size_t length) OVERRIDE; | |
50 | |
51 // When |in_failure_mode_| == true, the socket fails to send. | |
52 void SetFailureMode(bool in_failure_mode) { | |
53 in_failure_mode_ = in_failure_mode; | |
54 } | |
55 | |
56 private: | |
57 SharedBuffer* buffer_; | |
58 const OnSendCB on_send_cb_; | |
59 bool in_failure_mode_; | |
60 }; | |
61 | |
62 MockSyncSocket::MockSyncSocket(SharedBuffer* buffer) | |
63 : buffer_(buffer), in_failure_mode_(false) {} | |
64 | |
65 MockSyncSocket::MockSyncSocket(SharedBuffer* buffer, const OnSendCB& on_send_cb) | |
66 : buffer_(buffer), on_send_cb_(on_send_cb), in_failure_mode_(false) {} | |
67 | |
68 size_t MockSyncSocket::Send(const void* buffer, size_t length) { | |
69 if (in_failure_mode_) return 0; | |
70 uint8* b = static_cast<uint8*>(const_cast<void*>(buffer)); | |
71 for (size_t i = 0; i < length; i++, buffer_->length++) | |
72 buffer_->data[buffer_->start + buffer_->length] = b[i]; | |
73 on_send_cb_.Run(); | |
74 return length; | |
75 } | |
76 | |
77 size_t MockSyncSocket::Receive(void* buffer, size_t length) { | |
78 uint8* b = static_cast<uint8*>(const_cast<void*>(buffer)); | |
79 for (size_t i = buffer_->start; i < buffer_->length; i++, buffer_->start++) | |
80 b[i] = buffer_->data[buffer_->start]; | |
81 return length; | |
82 } | |
83 | |
84 //////////////////////////////////////////////////////////////////////////////// | |
85 | |
86 class FakeSpeechRecognizer { | |
87 public: | |
88 FakeSpeechRecognizer() : is_responsive_(true) {} | |
89 ~FakeSpeechRecognizer() {} | |
90 | |
91 void Initialize( | |
92 const blink::WebMediaStreamTrack& track, | |
93 const media::AudioParameters& sink_params, | |
94 const SpeechRecognitionAudioSourceProvider::OnErrorCB& on_error_cb); | |
95 | |
96 // TODO(burnik): Move from the recognizer to test. | |
97 SpeechRecognitionAudioSourceProvider* SourceProvider(); | |
98 | |
99 // Emulates a single iteraton of a thread receiving on the socket. | |
100 virtual void EmulateReceiveThreadLoopIteration(); | |
101 | |
102 // Used to simulate an unresponsive behaviour of the consumer. | |
103 void SimulateResponsiveness(bool is_responsive) { | |
104 is_responsive_ = is_responsive; | |
105 } | |
106 // Used to simulate a problem with sockets. | |
107 void SetFailureModeOnForeignSocket(bool in_failure_mode) { | |
108 DCHECK(foreign_socket_.get()); | |
109 foreign_socket_->SetFailureMode(in_failure_mode); | |
110 } | |
111 | |
112 uint32 buffer_index() { return *shared_buffer_index_; } | |
113 media::AudioBus* audio_bus() const { return audio_track_bus_.get(); } | |
114 | |
115 private: | |
116 bool is_responsive_; | |
117 // Shared memory for the audio and synchronization. | |
118 scoped_ptr<base::SharedMemory> shared_memory_; | |
119 | |
120 // Fake sockets shared buffer. | |
121 scoped_ptr<SharedBuffer> shared_buffer_; | |
122 scoped_ptr<MockSyncSocket> local_socket_; | |
123 scoped_ptr<MockSyncSocket> foreign_socket_; | |
124 | |
125 // Audio bus wrapping the shared memory from the renderer. | |
126 scoped_ptr<media::AudioBus> audio_track_bus_; | |
127 | |
128 uint32* shared_buffer_index_; | |
129 // Producer. TODO(burnik): this should be outside the recognizer. | |
130 scoped_ptr<SpeechRecognitionAudioSourceProvider> audio_source_provider_; | |
131 }; | |
132 | |
133 void FakeSpeechRecognizer::Initialize( | |
134 const blink::WebMediaStreamTrack& track, | |
135 const media::AudioParameters& sink_params, | |
136 const SpeechRecognitionAudioSourceProvider::OnErrorCB& on_error_cb) { | |
137 // Shared memory is allocated, mapped and shared. | |
138 uint32 shared_memory_size = sizeof(media::AudioInputBufferParameters) + | |
139 media::AudioBus::CalculateMemorySize(sink_params); | |
140 shared_memory_.reset(new base::SharedMemory()); | |
141 | |
142 ASSERT_TRUE(shared_memory_->CreateAndMapAnonymous(shared_memory_size)) | |
143 << "Failed to create shared memory"; | |
144 | |
145 base::SharedMemoryHandle foreign_memory_handle; | |
146 ASSERT_TRUE(shared_memory_->ShareToProcess(base::GetCurrentProcessHandle(), | |
147 &foreign_memory_handle)) | |
148 << "Failed to share memory"; | |
149 | |
150 media::AudioInputBuffer* buffer = | |
151 static_cast<media::AudioInputBuffer*>(shared_memory_->memory()); | |
152 audio_track_bus_ = media::AudioBus::WrapMemory(sink_params, buffer->audio); | |
153 | |
154 // Reference to the counter used to synchronize. | |
155 shared_buffer_index_ = &(buffer->params.size); | |
156 *shared_buffer_index_ = 0U; | |
157 | |
158 // Create a shared buffer for the |MockSyncSocket|s. | |
159 shared_buffer_.reset(new SharedBuffer()); | |
160 ASSERT_EQ(shared_buffer_->start, 0U); | |
161 ASSERT_EQ(shared_buffer_->length, 0U); | |
162 | |
163 // Local socket will receive signals from the producer. | |
164 local_socket_.reset(new MockSyncSocket(shared_buffer_.get())); | |
165 | |
166 // We automatically trigger a Receive when data is sent over the socket. | |
167 foreign_socket_.reset(new MockSyncSocket( | |
168 shared_buffer_.get(), | |
169 base::Bind(&FakeSpeechRecognizer::EmulateReceiveThreadLoopIteration, | |
170 base::Unretained(this)))); | |
171 | |
172 // This is usually done to pair the sockets. Here it's not effective. | |
173 base::SyncSocket::CreatePair(local_socket_.get(), foreign_socket_.get()); | |
174 | |
175 // Create the producer. TODO(burnik): move out of the recognizer. | |
176 audio_source_provider_.reset(new SpeechRecognitionAudioSourceProvider( | |
177 track, sink_params, foreign_memory_handle, foreign_socket_.get(), | |
178 on_error_cb)); | |
179 } | |
180 | |
181 // TODO(burnik): Remove from the recognizer. | |
182 SpeechRecognitionAudioSourceProvider* FakeSpeechRecognizer::SourceProvider() { | |
183 return audio_source_provider_.get(); | |
184 } | |
185 | |
186 // Emulates the receive on the socket. This would normally be done on a | |
187 // receiving thread's loop on the browser. | |
188 void FakeSpeechRecognizer::EmulateReceiveThreadLoopIteration() { | |
189 // When not responsive do nothing as if the process is busy. | |
190 if (!is_responsive_) return; | |
191 local_socket_->Receive(shared_buffer_index_, sizeof(*shared_buffer_index_)); | |
192 // Notify the producer that the audio buffer has been consumed. | |
193 (*shared_buffer_index_)++; | |
194 } | |
195 | |
196 //////////////////////////////////////////////////////////////////////////////// | |
197 | |
198 // Input audio format | |
199 static const media::AudioParameters::Format kInputFormat = | |
200 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
201 const media::ChannelLayout kInputChannelLayout = media::CHANNEL_LAYOUT_MONO; | |
202 const int kInputChannels = 1; | |
203 const int kInputSampleRate = 44100; | |
204 const int kInputBitsPerSample = 16; | |
205 const int kInputFramesPerBuffer = 441; | |
206 | |
207 // Output audio format | |
208 const media::AudioParameters::Format kOutputFormat = | |
209 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
210 const media::ChannelLayout kOutputChannelLayout = media::CHANNEL_LAYOUT_STEREO; | |
211 const int kOutputChannels = 2; | |
212 const int kOutputSampleRate = 16000; | |
213 const int kOutputBitsPerSample = 16; | |
214 const int kOutputFramesPerBuffer = 1600; | |
215 | |
216 // Minimal number of buffers which trigger a single socket transfer. | |
217 const size_t kBuffersPerNotification = | |
218 (kOutputFramesPerBuffer * kInputSampleRate) / | |
219 (kInputFramesPerBuffer * kOutputSampleRate); | |
220 | |
221 // Number of buffers which make the FIFO ready for consumption. | |
222 const size_t kBuffersForReadyFifo = | |
223 (kOutputFramesPerBuffer * kInputSampleRate) / kOutputSampleRate; | |
224 | |
225 //////////////////////////////////////////////////////////////////////////////// | |
226 | |
227 class SpeechRecognitionAudioSourceProviderTest : public testing::Test { | |
228 public: | |
229 SpeechRecognitionAudioSourceProviderTest() {} | |
230 | |
231 // Mock for error callback. | |
232 MOCK_METHOD1(ErrorCallback, | |
233 void(SpeechRecognitionAudioSourceProvider::ErrorState)); | |
234 | |
235 // testing::Test methods. | |
236 virtual void SetUp() OVERRIDE { | |
237 // Audio Environment setup. | |
238 source_params_.Reset(kInputFormat, kInputChannelLayout, kInputChannels, | |
239 kInputSampleRate, kInputBitsPerSample, | |
240 kInputFramesPerBuffer); | |
241 | |
242 sink_params_.Reset(kOutputFormat, kOutputChannelLayout, kOutputChannels, | |
243 kOutputSampleRate, kOutputBitsPerSample, | |
244 kOutputFramesPerBuffer); | |
245 | |
246 source_data_length_ = | |
247 source_params_.frames_per_buffer() * source_params_.channels(); | |
248 source_data_.reset(new int16[source_data_length_]); | |
249 | |
250 MockMediaConstraintFactory constraint_factory; | |
251 scoped_refptr<WebRtcAudioCapturer> capturer( | |
252 WebRtcAudioCapturer::CreateCapturer( | |
253 -1, StreamDeviceInfo(), | |
254 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); | |
255 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
256 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
257 native_track_ = new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL); | |
258 native_track_->OnSetFormat(source_params_); | |
259 | |
260 blink::WebMediaStreamSource audio_source; | |
261 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | |
262 blink::WebMediaStreamSource::TypeAudio, | |
263 base::UTF8ToUTF16("dummy_source_name")); | |
264 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), | |
265 audio_source); | |
266 blink_track_.setExtraData(native_track_); | |
267 | |
268 // Create the consumer. | |
269 recognizer_ = new FakeSpeechRecognizer(); | |
270 recognizer_->Initialize( | |
271 blink_track_, sink_params_, | |
272 base::Bind(&SpeechRecognitionAudioSourceProviderTest::ErrorCallback, | |
273 base::Unretained(this))); | |
274 | |
275 // Init the producer. | |
276 audio_source_provider_.reset(recognizer_->SourceProvider()); | |
277 } | |
278 | |
279 virtual void TearDown() OVERRIDE { blink_track_.reset(); } | |
280 | |
281 protected: | |
282 // TODO(burnik): Recheck steps and simplify method. Try reusing in |SetUp()|. | |
283 static blink::WebMediaStreamTrack CreateBlinkTrackWithMediaStreamType( | |
284 const MediaStreamType device_type) { | |
285 MockMediaConstraintFactory constraint_factory; | |
286 | |
287 MediaStreamSource::SourceStoppedCallback cb; | |
288 | |
289 StreamDeviceInfo device_info(device_type, "Mock audio device", | |
290 "mock_audio_device_id"); | |
291 WebRtcAudioDeviceImpl* device = new WebRtcAudioDeviceImpl(); | |
292 scoped_ptr<MediaStreamAudioSource> stream_audio_source( | |
293 new MediaStreamAudioSource(-1, device_info, cb, NULL)); | |
294 const blink::WebMediaConstraints constraints = | |
295 constraint_factory.CreateWebMediaConstraints(); | |
296 MockPeerConnectionDependencyFactory* factory = | |
297 new MockPeerConnectionDependencyFactory(); | |
298 scoped_refptr<webrtc::AudioSourceInterface> audio_source = | |
299 factory->CreateLocalAudioSource(new RTCMediaConstraints(constraints)); | |
300 scoped_refptr<WebRtcAudioCapturer> capturer( | |
301 WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, | |
302 device, stream_audio_source.get())); | |
303 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
304 WebRtcLocalAudioTrackAdapter::Create(std::string(), | |
305 audio_source.get())); | |
306 scoped_ptr<WebRtcLocalAudioTrack> native_track( | |
307 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | |
308 | |
309 blink::WebMediaStreamSource blink_audio_source; | |
310 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | |
311 blink::WebMediaStreamSource::TypeAudio, | |
312 base::UTF8ToUTF16("dummy_source_name")); | |
313 blink_audio_source.setExtraData(stream_audio_source.release()); | |
314 | |
315 blink::WebMediaStreamTrack blink_track; | |
316 blink_track.initialize(blink::WebString::fromUTF8("audio_track"), | |
317 blink_audio_source); | |
318 blink_track.setExtraData(native_track.release()); | |
319 | |
320 return blink_track; | |
321 } | |
322 | |
323 // Emulates an audio capture device capturing data from the source. | |
324 inline void CaptureAudio(const size_t buffers) { | |
325 DCHECK(native_track_); | |
326 for (size_t i = 0; i < buffers; ++i) | |
327 native_track_->Capture(source_data_.get(), | |
328 base::TimeDelta::FromMilliseconds(0), 1, false, | |
329 false); | |
330 } | |
331 | |
332 // Helper method to verify captured audio data has been consumed. | |
333 inline void AssertConsumedBuffers(const size_t buffer_index) { | |
334 ASSERT_EQ(buffer_index, recognizer_->buffer_index()); | |
335 } | |
336 | |
337 // Helper method to push audio data to producer and verify it was consumed. | |
338 inline void CaptureAudioAndAssertConsumedBuffers(const size_t buffers, | |
339 const size_t buffer_index) { | |
340 CaptureAudio(buffers); | |
341 AssertConsumedBuffers(buffer_index); | |
342 } | |
343 | |
344 protected: | |
345 // Producer. | |
346 scoped_ptr<SpeechRecognitionAudioSourceProvider> audio_source_provider_; | |
347 // Consumer. | |
348 FakeSpeechRecognizer* recognizer_; | |
349 // Audio related members. | |
350 size_t source_data_length_; | |
351 media::AudioParameters source_params_; | |
352 scoped_ptr<int16[]> source_data_; | |
353 size_t sink_data_length_; | |
354 media::AudioParameters sink_params_; | |
355 blink::WebMediaStreamTrack blink_track_; | |
356 WebRtcLocalAudioTrack* native_track_; | |
357 }; | |
358 | |
359 //////////////////////////////////////////////////////////////////////////////// | |
360 //////////////////////////////////////////////////////////////////////////////// | |
361 | |
362 TEST_F(SpeechRecognitionAudioSourceProviderTest, CheckAllowedAudioTrackType) { | |
363 typedef std::map<MediaStreamType, bool> AllowedAudioTrackSourceTypePolicy; | |
364 // This test must be aligned with the policy of allowed tracks. | |
365 AllowedAudioTrackSourceTypePolicy p; | |
366 p[MEDIA_NO_SERVICE] = false; | |
367 p[MEDIA_DEVICE_AUDIO_CAPTURE] = true; // Only one allowed for now. | |
368 p[MEDIA_DEVICE_VIDEO_CAPTURE] = false; | |
369 p[MEDIA_TAB_AUDIO_CAPTURE] = false; | |
370 p[MEDIA_TAB_VIDEO_CAPTURE] = false; | |
371 p[MEDIA_DESKTOP_VIDEO_CAPTURE] = false; | |
372 p[MEDIA_LOOPBACK_AUDIO_CAPTURE] = false; | |
373 p[MEDIA_DEVICE_AUDIO_OUTPUT] = false; | |
374 // Ensure this test gets updated along with |content::MediaStreamType| enum. | |
375 EXPECT_EQ(NUM_MEDIA_TYPES, p.size()); | |
376 // Check the the entire policy. | |
377 for (AllowedAudioTrackSourceTypePolicy::iterator it = p.begin(); | |
378 it != p.end(); ++it) { | |
379 ASSERT_EQ(it->second, | |
380 SpeechRecognitionAudioSourceProvider::IsAllowedAudioTrack( | |
381 CreateBlinkTrackWithMediaStreamType(it->first))); | |
382 } | |
383 } | |
384 | |
385 TEST_F(SpeechRecognitionAudioSourceProviderTest, RecognizerNotifiedOnSocket) { | |
386 AssertConsumedBuffers(0U); | |
387 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
388 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 2U); | |
389 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 3U); | |
390 } | |
391 | |
392 TEST_F(SpeechRecognitionAudioSourceProviderTest, AudioDataIsResampledOnSink) { | |
393 // fill audio input frames with 0,1,2,3,...,440 | |
394 for (size_t i = 0; i < source_data_length_; ++i) source_data_[i] = i; | |
395 | |
396 const size_t num_frames_to_test = 12; | |
397 int16 sink_data[kOutputFramesPerBuffer * kOutputChannels]; | |
398 media::AudioBus* sink_bus = recognizer_->audio_bus(); | |
399 | |
400 // Render the audio data from the recognizer. | |
401 sink_bus->ToInterleaved(sink_bus->frames(), | |
402 sink_params_.bits_per_sample() / 8, sink_data); | |
403 | |
404 // Test both channels are zeroed out before we trigger resampling. | |
405 for (size_t i = 0; i < num_frames_to_test; ++i) { | |
406 ASSERT_EQ(0, sink_data[i * 2]); | |
407 ASSERT_EQ(0, sink_data[i * 2 + 1]); | |
408 } | |
409 | |
410 // Trigger the source provider to resample the input data. | |
411 AssertConsumedBuffers(0U); | |
412 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
413 | |
414 // Render the audio data from the recognizer. | |
415 sink_bus->ToInterleaved(sink_bus->frames(), | |
416 sink_params_.bits_per_sample() / 8, sink_data); | |
417 | |
418 // Resampled data expected frames - based on |source_data_|. | |
419 // Note: these values also depend on input/output audio params. | |
420 const int16 expected_data[num_frames_to_test] = {0, 2, 5, 8, 11, 13, | |
421 16, 19, 22, 24, 27, 30}; | |
422 | |
423 // Test both channels have same resampled data. | |
424 for (size_t i = 0; i < num_frames_to_test; ++i) { | |
425 ASSERT_EQ(expected_data[i], sink_data[i * 2]); | |
426 ASSERT_EQ(expected_data[i], sink_data[i * 2 + 1]); | |
427 } | |
428 } | |
429 | |
430 TEST_F(SpeechRecognitionAudioSourceProviderTest, SyncSocketFailsSendingData) { | |
431 // (2) Start out with no problems. | |
432 AssertConsumedBuffers(0U); | |
433 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
434 | |
435 // (2) A failure occurs (socket cannot to send). | |
436 recognizer_->SetFailureModeOnForeignSocket(true); | |
437 EXPECT_CALL(*this, | |
438 ErrorCallback(SpeechRecognitionAudioSourceProvider::SEND_FAILED)) | |
439 .Times(1); | |
440 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
441 | |
442 // (3) Miraculasly recovered from the socket failure. | |
burnik
2014/09/12 12:09:12
* Miraculously
burnik
2014/09/15 15:00:07
Done.
| |
443 recognizer_->SetFailureModeOnForeignSocket(false); | |
444 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 2U); | |
445 } | |
446 | |
447 TEST_F(SpeechRecognitionAudioSourceProviderTest, PeerProcessGotUnresponsive) { | |
448 EXPECT_GT(kBuffersForReadyFifo, kBuffersPerNotification); | |
449 AssertConsumedBuffers(0U); | |
450 | |
451 // (1) We respond to audio packets as expected. | |
452 recognizer_->SimulateResponsiveness(true); | |
453 // First round of input has to have one additional buffer | |
burnik
2014/09/12 12:09:12
This comment is deprecated.
burnik
2014/09/15 15:00:07
Done.
| |
454 // to trigger processing. | |
455 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
456 | |
457 // (2) The recognizer on the browser becomes unresponsive. | |
458 recognizer_->SimulateResponsiveness(false); | |
459 EXPECT_CALL(*this, ErrorCallback( | |
460 SpeechRecognitionAudioSourceProvider::BUFFER_SYNC_LAG)) | |
461 .Times(testing::AtLeast(1)); | |
462 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
463 | |
464 // (2) The producer gets an overflow. | |
465 EXPECT_CALL( | |
466 *this, | |
467 ErrorCallback(SpeechRecognitionAudioSourceProvider::AUDIO_FIFO_OVERFLOW)) | |
468 .Times(testing::AtLeast(1)); | |
469 CaptureAudioAndAssertConsumedBuffers(kBuffersForReadyFifo, 1U); | |
470 } | |
471 | |
472 TEST_F(SpeechRecognitionAudioSourceProviderTest, OnReadyStateChangedOccured) { | |
473 AssertConsumedBuffers(0U); | |
474 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
475 EXPECT_CALL( | |
476 *this, ErrorCallback(SpeechRecognitionAudioSourceProvider::TRACK_STOPPED)) | |
477 .Times(1); | |
478 | |
479 native_track_->Stop(); | |
480 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
481 } | |
482 | |
483 } // namespace content | |
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