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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/speech_recognition_audio_sink.h" | 5 #include "content/renderer/media/speech_recognition_audio_sink.h" |
6 | 6 |
7 #include <stddef.h> | 7 #include <stddef.h> |
8 #include <stdint.h> | 8 #include <stdint.h> |
9 #include <string.h> | 9 #include <string.h> |
10 #include <utility> | 10 #include <utility> |
11 | 11 |
12 #include "base/bind.h" | 12 #include "base/bind.h" |
13 #include "base/macros.h" | 13 #include "base/macros.h" |
14 #include "base/strings/utf_string_conversions.h" | 14 #include "base/strings/utf_string_conversions.h" |
15 #include "content/renderer/media/media_stream_audio_source.h" | 15 #include "content/renderer/media/media_stream_audio_source.h" |
16 #include "content/renderer/media/mock_media_constraint_factory.h" | |
17 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 16 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
18 #include "content/renderer/media/webrtc_local_audio_track.h" | 17 #include "content/renderer/media/webrtc_local_audio_track.h" |
19 #include "media/audio/audio_parameters.h" | 18 #include "media/audio/audio_parameters.h" |
20 #include "media/base/audio_bus.h" | 19 #include "media/base/audio_bus.h" |
21 #include "testing/gmock/include/gmock/gmock.h" | 20 #include "testing/gmock/include/gmock/gmock.h" |
22 #include "testing/gtest/include/gtest/gtest.h" | 21 #include "testing/gtest/include/gtest/gtest.h" |
23 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 22 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
24 #include "third_party/WebKit/public/web/WebHeap.h" | 23 #include "third_party/WebKit/public/web/WebHeap.h" |
25 | 24 |
26 namespace { | 25 namespace { |
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268 | 267 |
269 // Mock callback expected to be called when the track is stopped. | 268 // Mock callback expected to be called when the track is stopped. |
270 MOCK_METHOD0(StoppedCallback, void()); | 269 MOCK_METHOD0(StoppedCallback, void()); |
271 | 270 |
272 protected: | 271 protected: |
273 // Prepares a blink track of a given MediaStreamType and attaches the native | 272 // Prepares a blink track of a given MediaStreamType and attaches the native |
274 // track which can be used to capture audio data and pass it to the producer. | 273 // track which can be used to capture audio data and pass it to the producer. |
275 static void PrepareBlinkTrackOfType( | 274 static void PrepareBlinkTrackOfType( |
276 const MediaStreamType device_type, | 275 const MediaStreamType device_type, |
277 blink::WebMediaStreamTrack* blink_track) { | 276 blink::WebMediaStreamTrack* blink_track) { |
278 StreamDeviceInfo device_info(device_type, "Mock device", | |
279 "mock_device_id"); | |
280 MockMediaConstraintFactory constraint_factory; | |
281 const blink::WebMediaConstraints constraints = | |
282 constraint_factory.CreateWebMediaConstraints(); | |
283 scoped_refptr<WebRtcAudioCapturer> capturer( | |
284 WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL, | |
285 NULL)); | |
286 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 277 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
287 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 278 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
288 scoped_ptr<WebRtcLocalAudioTrack> native_track( | 279 scoped_ptr<WebRtcLocalAudioTrack> native_track( |
289 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | 280 new WebRtcLocalAudioTrack(adapter.get())); |
290 blink::WebMediaStreamSource blink_audio_source; | 281 blink::WebMediaStreamSource blink_audio_source; |
291 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | 282 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), |
292 blink::WebMediaStreamSource::TypeAudio, | 283 blink::WebMediaStreamSource::TypeAudio, |
293 base::UTF8ToUTF16("dummy_source_name"), | 284 base::UTF8ToUTF16("dummy_source_name"), |
294 false /* remote */, true /* readonly */); | 285 false /* remote */, true /* readonly */); |
295 MediaStreamSource::SourceStoppedCallback cb; | 286 MediaStreamSource::SourceStoppedCallback cb; |
296 blink_audio_source.setExtraData( | 287 blink_audio_source.setExtraData(new MediaStreamAudioSource( |
297 new MediaStreamAudioSource(-1, device_info, cb, NULL)); | 288 -1, StreamDeviceInfo(device_type, "Mock device", "mock_device_id"), cb, |
| 289 nullptr)); |
298 blink_track->initialize(blink::WebString::fromUTF8("dummy_track"), | 290 blink_track->initialize(blink::WebString::fromUTF8("dummy_track"), |
299 blink_audio_source); | 291 blink_audio_source); |
300 blink_track->setExtraData(native_track.release()); | 292 blink_track->setExtraData(native_track.release()); |
301 } | 293 } |
302 | 294 |
303 // Emulates an audio capture device capturing data from the source. | 295 // Emulates an audio capture device capturing data from the source. |
304 inline void CaptureAudio(const uint32_t buffers) { | 296 inline void CaptureAudio(const uint32_t buffers) { |
305 for (uint32_t i = 0; i < buffers; ++i) { | 297 for (uint32_t i = 0; i < buffers; ++i) { |
306 const base::TimeTicks estimated_capture_time = first_frame_capture_time_ + | 298 const base::TimeTicks estimated_capture_time = first_frame_capture_time_ + |
307 (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) / | 299 (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) / |
308 source_params_.sample_rate()); | 300 source_params_.sample_rate()); |
309 native_track()->Capture(*source_bus_, estimated_capture_time, false); | 301 native_track()->Capture(*source_bus_, estimated_capture_time); |
310 sample_frames_captured_ += source_bus_->frames(); | 302 sample_frames_captured_ += source_bus_->frames(); |
311 } | 303 } |
312 } | 304 } |
313 | 305 |
314 // Used to simulate a problem with sockets. | 306 // Used to simulate a problem with sockets. |
315 void SetFailureModeOnForeignSocket(bool in_failure_mode) { | 307 void SetFailureModeOnForeignSocket(bool in_failure_mode) { |
316 recognizer()->sending_socket()->SetFailureMode(in_failure_mode); | 308 recognizer()->sending_socket()->SetFailureMode(in_failure_mode); |
317 } | 309 } |
318 | 310 |
319 // Helper method for verifying captured audio data has been consumed. | 311 // Helper method for verifying captured audio data has been consumed. |
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528 const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); | 520 const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); |
529 AssertConsumedBuffers(0U); | 521 AssertConsumedBuffers(0U); |
530 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); | 522 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); |
531 EXPECT_CALL(*this, StoppedCallback()).Times(1); | 523 EXPECT_CALL(*this, StoppedCallback()).Times(1); |
532 | 524 |
533 native_track()->Stop(); | 525 native_track()->Stop(); |
534 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); | 526 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); |
535 } | 527 } |
536 | 528 |
537 } // namespace content | 529 } // namespace content |
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