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Side by Side Diff: content/renderer/media/speech_recognition_audio_sink_unittest.cc

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: First attempt Created 4 years, 11 months ago
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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>
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285 NULL)); 285 NULL));
286 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( 286 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
287 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); 287 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
288 scoped_ptr<WebRtcLocalAudioTrack> native_track( 288 scoped_ptr<WebRtcLocalAudioTrack> native_track(
289 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); 289 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
290 blink::WebMediaStreamSource blink_audio_source; 290 blink::WebMediaStreamSource blink_audio_source;
291 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), 291 blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
292 blink::WebMediaStreamSource::TypeAudio, 292 blink::WebMediaStreamSource::TypeAudio,
293 base::UTF8ToUTF16("dummy_source_name"), 293 base::UTF8ToUTF16("dummy_source_name"),
294 false /* remote */, true /* readonly */); 294 false /* remote */, true /* readonly */);
295 MediaStreamSource::SourceStoppedCallback cb; 295 MediaStreamSource* const audio_source = new MediaStreamAudioSource();
296 blink_audio_source.setExtraData( 296 audio_source->SetDeviceInfo(device_info);
297 new MediaStreamAudioSource(-1, device_info, cb, NULL)); 297 blink_audio_source.setExtraData(audio_source); // Takes ownership.
298 blink_track->initialize(blink::WebString::fromUTF8("dummy_track"), 298 blink_track->initialize(blink::WebString::fromUTF8("dummy_track"),
299 blink_audio_source); 299 blink_audio_source);
300 blink_track->setExtraData(native_track.release()); 300 blink_track->setExtraData(native_track.release());
301 } 301 }
302 302
303 // Emulates an audio capture device capturing data from the source. 303 // Emulates an audio capture device capturing data from the source.
304 inline void CaptureAudio(const uint32_t buffers) { 304 inline void CaptureAudio(const uint32_t buffers) {
305 for (uint32_t i = 0; i < buffers; ++i) { 305 for (uint32_t i = 0; i < buffers; ++i) {
306 const base::TimeTicks estimated_capture_time = first_frame_capture_time_ + 306 const base::TimeTicks estimated_capture_time = first_frame_capture_time_ +
307 (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) / 307 (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) /
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528 const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600); 528 const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600);
529 AssertConsumedBuffers(0U); 529 AssertConsumedBuffers(0U);
530 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); 530 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U);
531 EXPECT_CALL(*this, StoppedCallback()).Times(1); 531 EXPECT_CALL(*this, StoppedCallback()).Times(1);
532 532
533 native_track()->Stop(); 533 native_track()->Stop();
534 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U); 534 CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U);
535 } 535 }
536 536
537 } // namespace content 537 } // namespace content
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