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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_source_provider.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 content { | |
18 | |
19 // Mocked out sockets used for Send/Receive. | |
20 // Data is written and read from a shared buffer used as a FIFO and there is | |
21 // no blocking. |OnSendCB| is used to trigger a |Receive| on the other socket. | |
22 class MockSyncSocket : public base::SyncSocket { | |
23 public: | |
24 // This allows for 2 requests in queue between the |MockSyncSocket|s. | |
25 static const int kSharedBufferSize = 8; | |
26 | |
27 // Buffer to be shared between two |MockSyncSocket|s. Allocated on heap. | |
28 struct SharedBuffer { | |
29 SharedBuffer() : start(0), length(0) {} | |
30 | |
31 uint8 data[kSharedBufferSize]; | |
32 size_t start; | |
33 size_t length; | |
34 }; | |
35 | |
36 // Callback used for pairing an A.Send() with B.Receieve() without blocking. | |
37 typedef base::Callback<void()> OnSendCB; | |
38 | |
39 explicit MockSyncSocket(SharedBuffer* shared_buffer) | |
40 : buffer_(shared_buffer), | |
41 in_failure_mode_(false) { } | |
42 | |
43 MockSyncSocket(SharedBuffer* shared_buffer, const OnSendCB& on_send_cb) | |
44 : buffer_(shared_buffer), | |
45 on_send_cb_(on_send_cb), | |
46 in_failure_mode_(false) { } | |
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 size_t MockSyncSocket::Send(const void* buffer, size_t length) { | |
63 if (in_failure_mode_) | |
64 return 0; | |
65 | |
66 uint8* b = static_cast<uint8*>(const_cast<void*>(buffer)); | |
67 for (size_t i = 0; i < length; i++, buffer_->length++) | |
burnik
2014/09/23 12:39:21
Changed to prefixed increment.
| |
68 buffer_->data[buffer_->start + buffer_->length] = b[i]; | |
69 | |
70 on_send_cb_.Run(); | |
71 return length; | |
72 } | |
73 | |
74 size_t MockSyncSocket::Receive(void* buffer, size_t length) { | |
75 uint8* b = static_cast<uint8*>(const_cast<void*>(buffer)); | |
76 for (size_t i = buffer_->start; i < buffer_->length; i++, buffer_->start++) | |
burnik
2014/09/23 12:39:20
Changed to prefixed increment.
| |
77 b[i] = buffer_->data[buffer_->start]; | |
78 | |
79 // Since buffer is used sequentially, we can reset the buffer indices here. | |
80 buffer_->start = buffer_->length = 0; | |
81 return length; | |
82 } | |
83 | |
84 //////////////////////////////////////////////////////////////////////////////// | |
henrika (OOO until Aug 14)
2014/09/22 08:02:19
Please remove these non-standard separators.
burnik
2014/09/22 09:17:36
Done.
no longer working on chromium
2014/09/23 10:09:13
Not done yet.
burnik
2014/09/23 12:39:21
Yes, done for next patchset as advertised.
| |
85 | |
86 class FakeSpeechRecognizer { | |
henrika (OOO until Aug 14)
2014/09/22 08:02:19
This looks like a very complex helper which is now
burnik
2014/09/22 09:17:36
This is the mock consumer. Unit tests focus on the
henrika (OOO until Aug 14)
2014/09/23 10:45:33
I am not saying it will fail but that is a large h
burnik
2014/09/23 12:39:21
Ok. I'll revisit existing unit tests to see if any
| |
87 public: | |
88 FakeSpeechRecognizer() : is_responsive_(true) { } | |
89 | |
90 void Initialize( | |
91 const blink::WebMediaStreamTrack& track, | |
92 const media::AudioParameters& sink_params, | |
93 base::SharedMemoryHandle* foreign_memory_handle) { | |
94 // Shared memory is allocated, mapped and shared. | |
95 uint32 shared_memory_size = | |
96 sizeof(media::AudioInputBufferParameters) + | |
97 media::AudioBus::CalculateMemorySize(sink_params); | |
98 shared_memory_.reset(new base::SharedMemory()); | |
99 ASSERT_TRUE(shared_memory_->CreateAndMapAnonymous(shared_memory_size)); | |
100 ASSERT_TRUE(shared_memory_->ShareToProcess(base::GetCurrentProcessHandle(), | |
101 foreign_memory_handle)); | |
102 | |
103 // Wrap the shared memory for the audio bus. | |
104 media::AudioInputBuffer* buffer = | |
105 static_cast<media::AudioInputBuffer*>(shared_memory_->memory()); | |
106 audio_track_bus_ = media::AudioBus::WrapMemory(sink_params, buffer->audio); | |
107 | |
108 // Reference to the counter used to synchronize. | |
109 buffer_index_ = &(buffer->params.size); | |
110 *buffer_index_ = 0U; | |
111 | |
112 // Create a shared buffer for the |MockSyncSocket|s. | |
113 shared_buffer_.reset(new MockSyncSocket::SharedBuffer()); | |
114 | |
115 // Local socket will receive signals from the producer. | |
116 local_socket_.reset(new MockSyncSocket(shared_buffer_.get())); | |
117 | |
118 // We automatically trigger a Receive when data is sent over the socket. | |
119 foreign_socket_ = new MockSyncSocket( | |
120 shared_buffer_.get(), | |
121 base::Bind(&FakeSpeechRecognizer::EmulateReceiveThreadLoopIteration, | |
122 base::Unretained(this))); | |
123 | |
124 // This is usually done to pair the sockets. Here it's not effective. | |
125 base::SyncSocket::CreatePair(local_socket_.get(), foreign_socket_); | |
126 } | |
127 | |
128 // Emulates a single iteraton of a thread receiving on the socket. | |
129 // This would normally be done on a receiving thread's task on the browser. | |
130 void EmulateReceiveThreadLoopIteration() { | |
131 // When not responsive do nothing as if the process is busy. | |
132 if (!is_responsive_) | |
133 return; | |
134 | |
135 local_socket_->Receive(buffer_index_, sizeof(*buffer_index_)); | |
136 // Notify the producer that the audio buffer has been consumed. | |
137 (*buffer_index_)++; | |
no longer working on chromium
2014/09/23 10:09:13
++(*buffer_index_)
burnik
2014/09/23 12:39:21
Done.
| |
138 } | |
139 | |
140 // Used to simulate an unresponsive behaviour of the consumer. | |
141 void SimulateResponsiveness(bool is_responsive) { | |
142 is_responsive_ = is_responsive; | |
143 } | |
144 | |
145 MockSyncSocket* foreign_socket() { return foreign_socket_; } | |
146 media::AudioBus* audio_bus() const { return audio_track_bus_.get(); } | |
147 uint32 buffer_index() { return *buffer_index_; } | |
148 | |
149 private: | |
150 bool is_responsive_; | |
151 | |
152 // Shared memory for the audio and synchronization. | |
153 scoped_ptr<base::SharedMemory> shared_memory_; | |
154 | |
155 // Fake sockets shared buffer. | |
156 scoped_ptr<MockSyncSocket::SharedBuffer> shared_buffer_; | |
157 scoped_ptr<MockSyncSocket> local_socket_; | |
158 MockSyncSocket* foreign_socket_; | |
no longer working on chromium
2014/09/23 10:09:13
why this is a raw pointer?
burnik
2014/09/23 12:39:21
It is owned by the recognizer and destroyed there.
| |
159 | |
160 // Audio bus wrapping the shared memory from the renderer. | |
161 scoped_ptr<media::AudioBus> audio_track_bus_; | |
162 | |
163 // Used for synchronization of sent/received buffers. | |
164 uint32* buffer_index_; | |
165 }; | |
166 | |
167 //////////////////////////////////////////////////////////////////////////////// | |
henrika (OOO until Aug 14)
2014/09/22 08:02:19
remove
burnik
2014/09/22 09:17:36
Done.
| |
168 | |
169 namespace { | |
170 | |
171 // Supported speech recognition audio parameters. | |
172 const int kSpeechRecognitionSampleRate = 16000; | |
173 const int kSpeechRecognitionFramesPerBuffer = 1600; | |
174 | |
175 // Input audio format. | |
176 const media::AudioParameters::Format kInputFormat = | |
177 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
178 const media::ChannelLayout kInputChannelLayout = media::CHANNEL_LAYOUT_MONO; | |
179 const int kInputChannels = 1; | |
180 const int kInputBitsPerSample = 16; | |
181 | |
182 // Output audio format. | |
183 const media::AudioParameters::Format kOutputFormat = | |
184 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
185 const media::ChannelLayout kOutputChannelLayout = media::CHANNEL_LAYOUT_STEREO; | |
186 const int kOutputChannels = 2; | |
187 const int kOutputBitsPerSample = 16; | |
188 | |
189 } // namespace | |
190 | |
191 //////////////////////////////////////////////////////////////////////////////// | |
192 | |
193 class SpeechRecognitionAudioSourceProviderTest : public testing::Test { | |
194 public: | |
195 SpeechRecognitionAudioSourceProviderTest() { } | |
196 | |
197 // Initializes the producer and consumer with specified audio parameters. | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
Can you elaborate on what a producer and consumer
burnik
2014/09/22 09:17:36
Yes. It's explained on lines 228 - 238.
| |
198 // Returns the minimal number of input audio buffers which need to be captured | |
199 // before they get sent to the consumer. | |
200 uint32 Initialize(int input_sample_rate, | |
201 int input_frames_per_buffer, | |
202 int output_sample_rate, | |
203 int output_frames_per_buffer) { | |
204 // Audio Environment setup. | |
205 source_params_.Reset(kInputFormat, | |
206 kInputChannelLayout, | |
207 kInputChannels, | |
208 input_sample_rate, | |
209 kInputBitsPerSample, | |
210 input_frames_per_buffer); | |
211 sink_params_.Reset(kOutputFormat, | |
212 kOutputChannelLayout, | |
213 kOutputChannels, | |
214 output_sample_rate, | |
215 kOutputBitsPerSample, | |
216 output_frames_per_buffer); | |
217 source_data_.reset(new int16[input_frames_per_buffer * kInputChannels]); | |
218 | |
219 // Prepare the track and audio source. | |
220 blink::WebMediaStreamTrack blink_track; | |
221 PrepareBlinkTrackOfType(MEDIA_DEVICE_AUDIO_CAPTURE, &blink_track); | |
222 | |
223 // Get the native track from the blink track and initialize. | |
224 native_track_ = | |
225 static_cast<WebRtcLocalAudioTrack*>(blink_track.extraData()); | |
226 native_track_->OnSetFormat(source_params_); | |
227 | |
228 // Create and initialize the consumer. | |
229 recognizer_.reset(new FakeSpeechRecognizer()); | |
230 base::SharedMemoryHandle foreign_memory_handle; | |
231 recognizer_->Initialize(blink_track, sink_params_, &foreign_memory_handle); | |
232 | |
233 // Create the producer. | |
234 audio_source_provider_.reset(new SpeechRecognitionAudioSourceProvider( | |
235 blink_track, sink_params_, foreign_memory_handle, | |
236 recognizer_->foreign_socket(), | |
237 base::Bind(&SpeechRecognitionAudioSourceProviderTest::StoppedCallback, | |
238 base::Unretained(this)))); | |
239 | |
240 // Return number of buffers needed to trigger resampling and consumption. | |
241 return static_cast<uint32>(std::ceil( | |
242 static_cast<double>(output_frames_per_buffer * input_sample_rate) / | |
243 (input_frames_per_buffer * output_sample_rate))); | |
244 } | |
245 | |
246 // Mock callback expected to be called when the track is stopped. | |
247 MOCK_METHOD0(StoppedCallback, void()); | |
248 | |
249 protected: | |
250 static void PrepareBlinkTrackOfType( | |
251 const MediaStreamType device_type, | |
252 blink::WebMediaStreamTrack* blink_track) { | |
253 // Device info. | |
254 StreamDeviceInfo device_info(device_type, "Mock audio device", | |
255 "mock_audio_device_id"); | |
256 | |
257 // Constraints. | |
258 MockMediaConstraintFactory constraint_factory; | |
259 const blink::WebMediaConstraints constraints = | |
260 constraint_factory.CreateWebMediaConstraints(); | |
261 | |
262 // Capturer. | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
These comments does not add much. Please explain w
burnik
2014/09/22 09:17:36
All these comments are now removed.
| |
263 scoped_refptr<WebRtcAudioCapturer> capturer( | |
264 WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL, | |
265 NULL)); | |
266 | |
267 // Adapter. | |
268 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
269 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
270 | |
271 // Native track. | |
272 scoped_ptr<WebRtcLocalAudioTrack> native_track( | |
273 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | |
274 | |
275 // Blink audio source. | |
276 blink::WebMediaStreamSource blink_audio_source; | |
277 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | |
278 blink::WebMediaStreamSource::TypeAudio, | |
279 base::UTF8ToUTF16("dummy_source_name")); | |
280 MediaStreamSource::SourceStoppedCallback cb; | |
281 blink_audio_source.setExtraData( | |
282 new MediaStreamAudioSource(-1, device_info, cb, NULL)); | |
283 | |
284 // Blink track. | |
285 blink_track->initialize(blink::WebString::fromUTF8("dummy_audio_track"), | |
286 blink_audio_source); | |
287 blink_track->setExtraData(native_track.release()); | |
288 } | |
289 | |
290 // Emulates an audio capture device capturing data from the source. | |
291 inline void CaptureAudio(const uint32 buffers) { | |
292 for (uint32 i = 0; i < buffers; ++i) | |
293 native_track_->Capture(source_data_.get(), | |
294 base::TimeDelta::FromMilliseconds(0), 1, false, | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
FromMilliseconds(0)?
burnik
2014/09/22 09:17:36
Yes, no delay is required in the unit test.
| |
295 false); | |
296 } | |
297 | |
298 // Used to simulate a problem with sockets. | |
299 void SetFailureModeOnForeignSocket(bool in_failure_mode) { | |
300 recognizer_->foreign_socket()->SetFailureMode(in_failure_mode); | |
301 } | |
302 | |
303 // Helper method for verifying captured audio data has been consumed. | |
304 inline void AssertConsumedBuffers(const uint32 buffer_index) { | |
305 ASSERT_EQ(buffer_index, recognizer_->buffer_index()); | |
306 } | |
307 | |
308 // Helper method for providing audio data to producer and verifying it was | |
309 // consumed on the recognizer. | |
310 inline void CaptureAudioAndAssertConsumedBuffers(const uint32 buffers, | |
311 const uint32 buffer_index) { | |
312 CaptureAudio(buffers); | |
313 AssertConsumedBuffers(buffer_index); | |
314 } | |
315 | |
316 // Helper method to capture and assert consumption at different sample rates | |
317 // and audio buffer sizes. | |
318 inline void AssertConsumptionForAudioParameters( | |
319 const int input_sample_rate, | |
320 const int input_frames_per_buffer, | |
321 const int output_sample_rate, | |
322 const int output_frames_per_buffer, | |
323 const uint32 consumptions) { | |
324 const uint32 kBuffersPerNotification = | |
325 Initialize(input_sample_rate, input_frames_per_buffer, | |
326 output_sample_rate, output_frames_per_buffer); | |
327 AssertConsumedBuffers(0U); | |
328 | |
329 for (uint32 i = 1U; i <= consumptions; ++i) { | |
330 CaptureAudio(kBuffersPerNotification); | |
331 ASSERT_EQ(i, recognizer_->buffer_index()) | |
332 << "Tested at rates: " | |
333 << "In(" << input_sample_rate << ", " << input_frames_per_buffer | |
334 << ") " | |
335 << "Out(" << output_sample_rate << ", " << output_frames_per_buffer | |
336 << ")"; | |
337 } | |
338 } | |
339 | |
340 // Producer. | |
341 scoped_ptr<SpeechRecognitionAudioSourceProvider> audio_source_provider_; | |
342 | |
343 // Consumer. | |
344 scoped_ptr<FakeSpeechRecognizer> recognizer_; | |
345 | |
346 // Audio related members. | |
347 scoped_ptr<int16[]> source_data_; | |
348 media::AudioParameters source_params_; | |
349 media::AudioParameters sink_params_; | |
350 WebRtcLocalAudioTrack* native_track_; | |
351 }; | |
352 | |
353 //////////////////////////////////////////////////////////////////////////////// | |
354 | |
355 TEST_F(SpeechRecognitionAudioSourceProviderTest, CheckIsSupportedAudioTrack) { | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
Could you make the name more clear? CheckIsSupport
burnik
2014/09/22 09:17:37
Added comment above test.
| |
356 typedef std::map<MediaStreamType, bool> SupportedTrackPolicy; | |
357 | |
358 // This test must be aligned with the policy of supported tracks. | |
359 SupportedTrackPolicy p; | |
360 p[MEDIA_NO_SERVICE] = false; | |
361 p[MEDIA_DEVICE_AUDIO_CAPTURE] = true; // The only one supported for now. | |
362 p[MEDIA_DEVICE_VIDEO_CAPTURE] = false; | |
363 p[MEDIA_TAB_AUDIO_CAPTURE] = false; | |
364 p[MEDIA_TAB_VIDEO_CAPTURE] = false; | |
365 p[MEDIA_DESKTOP_VIDEO_CAPTURE] = false; | |
366 p[MEDIA_LOOPBACK_AUDIO_CAPTURE] = false; | |
367 p[MEDIA_DEVICE_AUDIO_OUTPUT] = false; | |
368 | |
369 // Ensure this test gets updated along with |content::MediaStreamType| enum. | |
370 EXPECT_EQ(NUM_MEDIA_TYPES, p.size()); | |
371 | |
372 // Check the the entire policy. | |
373 for (SupportedTrackPolicy::iterator it = p.begin(); it != p.end(); ++it) { | |
374 blink::WebMediaStreamTrack blink_track; | |
375 PrepareBlinkTrackOfType(it->first, &blink_track); | |
376 ASSERT_EQ( | |
377 it->second, | |
378 SpeechRecognitionAudioSourceProvider::IsSupportedTrack(blink_track)); | |
379 } | |
380 } | |
381 | |
382 TEST_F(SpeechRecognitionAudioSourceProviderTest, RecognizerNotifiedOnSocket) { | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
Please add some lines of comments above each test
burnik
2014/09/22 09:17:37
Done.
| |
383 const size_t kNumAudioParamTuples = 22; | |
384 const int kAudioParams[kNumAudioParamTuples][2] = { | |
385 {8000, 80}, {8000, 800}, {16000, 160}, {16000, 1600}, | |
386 {32000, 320}, {32000, 3200}, {44100, 441}, {44100, 4410}, | |
387 {48000, 480}, {48000, 4800}, {96000, 960}, {96000, 9600}, | |
388 {11025, 111}, {11025, 1103}, {22050, 221}, {22050, 2205}, | |
389 {88200, 882}, {88200, 8820}, {176400, 1764}, {176400, 17640}, | |
390 {192000, 1920}, {192000, 19200}}; | |
391 | |
392 // Check all listed tuples of input sample rates and buffers sizes. | |
393 for (size_t i = 0; i < kNumAudioParamTuples; ++i) { | |
394 AssertConsumptionForAudioParameters( | |
395 kAudioParams[i][0], kAudioParams[i][1], | |
396 kSpeechRecognitionSampleRate, kSpeechRecognitionFramesPerBuffer, 3U); | |
397 } | |
398 } | |
399 | |
400 TEST_F(SpeechRecognitionAudioSourceProviderTest, AudioDataIsResampledOnSink) { | |
henrika (OOO until Aug 14)
2014/09/22 08:02:19
Lots of hardcoded values in this test. Makes it di
burnik
2014/09/22 09:17:37
Added more comments.
I don't test that the resampl
| |
401 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); | |
402 | |
403 // Fill audio input frames with 0, 1, 2, 3, ..., 440. | |
404 const uint32 source_data_length = 441 * kInputChannels; | |
405 for (uint32 i = 0; i < source_data_length; ++i) | |
406 source_data_[i] = i; | |
407 | |
408 const uint32 num_frames_to_test = 12; | |
409 const uint32 sink_data_length = 1600 * kOutputChannels; | |
410 int16 sink_data[sink_data_length]; | |
411 media::AudioBus* sink_bus = recognizer_->audio_bus(); | |
412 | |
413 // Render the audio data from the recognizer. | |
414 sink_bus->ToInterleaved(sink_bus->frames(), | |
415 sink_params_.bits_per_sample() / 8, sink_data); | |
416 | |
417 // Test both channels are zeroed out before we trigger resampling. | |
418 for (uint32 i = 0; i < num_frames_to_test; ++i) { | |
419 ASSERT_EQ(0, sink_data[i * 2]); | |
420 ASSERT_EQ(0, sink_data[i * 2 + 1]); | |
421 } | |
422 | |
423 // Trigger the source provider to resample the input data. | |
424 AssertConsumedBuffers(0U); | |
425 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
426 | |
427 // Render the audio data from the recognizer. | |
428 sink_bus->ToInterleaved(sink_bus->frames(), | |
429 sink_params_.bits_per_sample() / 8, sink_data); | |
430 | |
431 // Resampled data expected frames - based on |source_data_|. | |
432 const int16 expected_data[num_frames_to_test] = {0, 2, 5, 8, 11, 13, | |
433 16, 19, 22, 24, 27, 30}; | |
434 | |
435 // Test both channels have same resampled data. | |
436 for (uint32 i = 0; i < num_frames_to_test; ++i) { | |
437 ASSERT_EQ(expected_data[i], sink_data[i * 2]); | |
438 ASSERT_EQ(expected_data[i], sink_data[i * 2 + 1]); | |
439 } | |
440 } | |
441 | |
442 TEST_F(SpeechRecognitionAudioSourceProviderTest, SyncSocketFailsSendingData) { | |
443 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); | |
444 // (1) Start with no problems on the socket. | |
henrika (OOO until Aug 14)
2014/09/22 08:02:18
Remove (1) and (2)
burnik
2014/09/22 09:17:37
Done.
| |
445 AssertConsumedBuffers(0U); | |
446 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
447 | |
448 // (2) A failure occurs (socket cannot to send). | |
449 SetFailureModeOnForeignSocket(true); | |
450 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
451 } | |
452 | |
453 TEST_F(SpeechRecognitionAudioSourceProviderTest, OnReadyStateChangedOccured) { | |
454 const uint32 kBuffersPerNotification = Initialize(44100, 441, 16000, 1600); | |
455 AssertConsumedBuffers(0U); | |
456 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
457 EXPECT_CALL(*this, StoppedCallback()).Times(1); | |
458 | |
459 native_track_->Stop(); | |
460 CaptureAudioAndAssertConsumedBuffers(kBuffersPerNotification, 1U); | |
461 } | |
462 | |
463 } // namespace content | |
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