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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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 "media/audio/win/audio_low_latency_input_win.h" | 5 #include "media/audio/win/audio_low_latency_input_win.h" |
6 | 6 |
7 #include <memory> | 7 #include <memory> |
8 | 8 |
9 #include "base/logging.h" | 9 #include "base/logging.h" |
10 #include "base/metrics/histogram_macros.h" | 10 #include "base/metrics/histogram_macros.h" |
11 #include "base/strings/utf_string_conversions.h" | 11 #include "base/strings/utf_string_conversions.h" |
12 #include "base/trace_event/trace_event.h" | 12 #include "base/trace_event/trace_event.h" |
13 #include "media/audio/audio_device_description.h" | 13 #include "media/audio/audio_device_description.h" |
14 #include "media/audio/win/audio_manager_win.h" | 14 #include "media/audio/win/audio_manager_win.h" |
15 #include "media/audio/win/avrt_wrapper_win.h" | 15 #include "media/audio/win/avrt_wrapper_win.h" |
16 #include "media/audio/win/core_audio_util_win.h" | 16 #include "media/audio/win/core_audio_util_win.h" |
17 #include "media/base/audio_bus.h" | 17 #include "media/base/audio_bus.h" |
18 #include "media/base/channel_layout.h" | |
19 #include "media/base/limits.h" | |
20 #include "media/base/multi_channel_resampler.h" | |
18 | 21 |
19 using base::win::ScopedComPtr; | 22 using base::win::ScopedComPtr; |
20 using base::win::ScopedCOMInitializer; | 23 using base::win::ScopedCOMInitializer; |
21 | 24 |
22 namespace media { | 25 namespace media { |
23 | 26 |
24 WASAPIAudioInputStream::WASAPIAudioInputStream(AudioManagerWin* manager, | 27 WASAPIAudioInputStream::WASAPIAudioInputStream(AudioManagerWin* manager, |
25 const AudioParameters& params, | 28 const AudioParameters& params, |
26 const std::string& device_id) | 29 const std::string& device_id) |
27 : manager_(manager), | 30 : manager_(manager), |
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116 // set during construction. | 119 // set during construction. |
117 if (!DesiredFormatIsSupported()) { | 120 if (!DesiredFormatIsSupported()) { |
118 open_result_ = OPEN_RESULT_FORMAT_NOT_SUPPORTED; | 121 open_result_ = OPEN_RESULT_FORMAT_NOT_SUPPORTED; |
119 ReportOpenResult(); | 122 ReportOpenResult(); |
120 return false; | 123 return false; |
121 } | 124 } |
122 | 125 |
123 // Initialize the audio stream between the client and the device using | 126 // Initialize the audio stream between the client and the device using |
124 // shared mode and a lowest possible glitch-free latency. | 127 // shared mode and a lowest possible glitch-free latency. |
125 hr = InitializeAudioEngine(); | 128 hr = InitializeAudioEngine(); |
129 if (SUCCEEDED(hr) && converter_) | |
130 open_result_ = OPEN_RESULT_OK_WITH_RESAMPLING; | |
126 ReportOpenResult(); // Report before we assign a value to |opened_|. | 131 ReportOpenResult(); // Report before we assign a value to |opened_|. |
127 opened_ = SUCCEEDED(hr); | 132 opened_ = SUCCEEDED(hr); |
128 DCHECK(open_result_ == OPEN_RESULT_OK || !opened_); | |
129 | 133 |
130 return opened_; | 134 return opened_; |
131 } | 135 } |
132 | 136 |
133 void WASAPIAudioInputStream::Start(AudioInputCallback* callback) { | 137 void WASAPIAudioInputStream::Start(AudioInputCallback* callback) { |
134 DCHECK(CalledOnValidThread()); | 138 DCHECK(CalledOnValidThread()); |
135 DCHECK(callback); | 139 DCHECK(callback); |
136 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; | 140 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; |
137 if (!opened_) | 141 if (!opened_) |
138 return; | 142 return; |
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220 started_ = false; | 224 started_ = false; |
221 sink_ = NULL; | 225 sink_ = NULL; |
222 } | 226 } |
223 | 227 |
224 void WASAPIAudioInputStream::Close() { | 228 void WASAPIAudioInputStream::Close() { |
225 DVLOG(1) << "WASAPIAudioInputStream::Close()"; | 229 DVLOG(1) << "WASAPIAudioInputStream::Close()"; |
226 // It is valid to call Close() before calling open or Start(). | 230 // It is valid to call Close() before calling open or Start(). |
227 // It is also valid to call Close() after Start() has been called. | 231 // It is also valid to call Close() after Start() has been called. |
228 Stop(); | 232 Stop(); |
229 | 233 |
234 if (converter_) | |
235 converter_->RemoveInput(this); | |
236 | |
230 // Inform the audio manager that we have been closed. This will cause our | 237 // Inform the audio manager that we have been closed. This will cause our |
231 // destruction. | 238 // destruction. |
232 manager_->ReleaseInputStream(this); | 239 manager_->ReleaseInputStream(this); |
233 } | 240 } |
234 | 241 |
235 double WASAPIAudioInputStream::GetMaxVolume() { | 242 double WASAPIAudioInputStream::GetMaxVolume() { |
236 // Verify that Open() has been called succesfully, to ensure that an audio | 243 // Verify that Open() has been called succesfully, to ensure that an audio |
237 // session exists and that an ISimpleAudioVolume interface has been created. | 244 // session exists and that an ISimpleAudioVolume interface has been created. |
238 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; | 245 DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully"; |
239 if (!opened_) | 246 if (!opened_) |
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417 // each time SetVolume() is called through IPC by the render-side AGC. | 424 // each time SetVolume() is called through IPC by the render-side AGC. |
418 GetAgcVolume(&volume); | 425 GetAgcVolume(&volume); |
419 | 426 |
420 // Deliver captured data to the registered consumer using a packet | 427 // Deliver captured data to the registered consumer using a packet |
421 // size which was specified at construction. | 428 // size which was specified at construction. |
422 uint32_t delay_frames = static_cast<uint32_t>(audio_delay_frames + 0.5); | 429 uint32_t delay_frames = static_cast<uint32_t>(audio_delay_frames + 0.5); |
423 while (buffer_frame_index >= packet_size_frames_) { | 430 while (buffer_frame_index >= packet_size_frames_) { |
424 // Copy data to audio bus to match the OnData interface. | 431 // Copy data to audio bus to match the OnData interface. |
425 uint8_t* audio_data = | 432 uint8_t* audio_data = |
426 reinterpret_cast<uint8_t*>(capture_buffer.get()); | 433 reinterpret_cast<uint8_t*>(capture_buffer.get()); |
427 audio_bus_->FromInterleaved(audio_data, audio_bus_->frames(), | 434 |
428 format_.wBitsPerSample / 8); | 435 if (converter_) { |
436 convert_bus_->FromInterleaved(audio_data, convert_bus_->frames(), | |
437 format_.wBitsPerSample / 8); | |
438 data_was_converted_ = false; | |
439 converter_->ConvertWithDelay(delay_frames, audio_bus_.get()); | |
440 if (!data_was_converted_) { | |
441 LOG(ERROR) << "Failed to convert enough samples."; | |
DaleCurtis
2017/02/16 02:03:36
Won't this trample whatever you have in audio_bus_
tommi (sloooow) - chröme
2017/02/16 21:59:46
Yes indeed it would. I wasn't able to repro this a
| |
442 converter_->ConvertWithDelay(delay_frames, audio_bus_.get()); | |
443 } | |
444 DCHECK(data_was_converted_); | |
445 } else { | |
446 audio_bus_->FromInterleaved(audio_data, audio_bus_->frames(), | |
447 format_.wBitsPerSample / 8); | |
448 } | |
429 | 449 |
430 // Deliver data packet, delay estimation and volume level to | 450 // Deliver data packet, delay estimation and volume level to |
431 // the user. | 451 // the user. |
432 sink_->OnData(this, audio_bus_.get(), delay_frames * frame_size_, | 452 sink_->OnData(this, audio_bus_.get(), delay_frames * frame_size_, |
433 volume); | 453 volume); |
434 | 454 |
435 // Store parts of the recorded data which can't be delivered | 455 // Store parts of the recorded data which can't be delivered |
436 // using the current packet size. The stored section will be used | 456 // using the current packet size. The stored section will be used |
437 // either in the next while-loop iteration or in the next | 457 // either in the next while-loop iteration or in the next |
438 // capture event. | 458 // capture event. |
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588 // internal processing. However, the format for an application stream | 608 // internal processing. However, the format for an application stream |
589 // typically must have the same number of channels and the same sample | 609 // typically must have the same number of channels and the same sample |
590 // rate as the stream format used by the device. | 610 // rate as the stream format used by the device. |
591 // Many audio devices support both PCM and non-PCM stream formats. However, | 611 // Many audio devices support both PCM and non-PCM stream formats. However, |
592 // the audio engine can mix only PCM streams. | 612 // the audio engine can mix only PCM streams. |
593 base::win::ScopedCoMem<WAVEFORMATEX> closest_match; | 613 base::win::ScopedCoMem<WAVEFORMATEX> closest_match; |
594 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, | 614 HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, |
595 &format_, &closest_match); | 615 &format_, &closest_match); |
596 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported " | 616 DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported " |
597 << "but a closest match exists."; | 617 << "but a closest match exists."; |
618 if (hr == S_FALSE && | |
619 closest_match->nSamplesPerSec >= limits::kMinSampleRate && | |
620 closest_match->nSamplesPerSec <= limits::kMaxSampleRate) { | |
621 // We want a 1:1 ratio between the buffers we get and the buffers we | |
622 // give to OnData so that each buffer we receive from the OS can be directly | |
623 // resampled to a buffer that matches with what the client asked for. | |
624 const double buffer_ratio = | |
625 format_.nSamplesPerSec / static_cast<double>(audio_bus_->frames()); | |
626 const size_t new_frames_per_buffer = | |
627 static_cast<size_t>(closest_match->nSamplesPerSec / buffer_ratio); | |
628 | |
629 const AudioParameters input( | |
630 AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
631 GuessChannelLayout(closest_match->nChannels), | |
DaleCurtis
2017/02/16 02:03:36
Need to check the result of this that it's not CHA
| |
632 closest_match->nSamplesPerSec, | |
633 // We need to be careful here to not pick the closest wBitsPerSample | |
DaleCurtis
2017/02/16 02:03:36
I think this statement is false, or at least shoul
tommi (sloooow) - chröme
2017/02/16 21:59:46
Oh interesting, thanks for pointing that out (and
| |
634 // match as we need to use the PCM format (which might not be what | |
635 // closeest_match->wFormat is) and the internal resampler doesn't | |
636 // support all formats we might get here. So, we stick to the | |
637 // wBitsPerSample that was asked for originally (most likely 16). | |
638 format_.wBitsPerSample, new_frames_per_buffer); | |
639 | |
640 const AudioParameters output(AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
641 GuessChannelLayout(format_.nChannels), | |
642 format_.nSamplesPerSec, format_.wBitsPerSample, | |
643 audio_bus_->frames()); | |
644 | |
645 converter_.reset(new AudioConverter(input, output, false)); | |
646 converter_->AddInput(this); | |
647 converter_->PrimeWithSilence(); | |
648 convert_bus_ = AudioBus::Create(input); | |
649 | |
650 // Now change the format we're going to ask for to better match with what | |
651 // the OS can provide. If we succeed in opening the stream with these | |
652 // params, we can take care of the required resampling. | |
653 format_.nSamplesPerSec = closest_match->nSamplesPerSec; | |
654 format_.nChannels = closest_match->nChannels; | |
655 format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels; | |
656 format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign; | |
657 | |
658 // Update our packet size assumptions based on the new format. | |
659 const auto new_bytes_per_buffer = convert_bus_->frames() * | |
660 format_.nChannels * | |
661 (format_.wBitsPerSample / 8); | |
662 packet_size_frames_ = new_bytes_per_buffer / format_.nBlockAlign; | |
663 packet_size_bytes_ = new_bytes_per_buffer; | |
664 frame_size_ = format_.nBlockAlign; | |
665 ms_to_frame_count_ = static_cast<double>(format_.nSamplesPerSec) / 1000.0; | |
666 | |
667 // Indicate that we're good to go with a close match. | |
668 hr = S_OK; | |
669 } | |
670 | |
598 return (hr == S_OK); | 671 return (hr == S_OK); |
599 } | 672 } |
600 | 673 |
601 HRESULT WASAPIAudioInputStream::InitializeAudioEngine() { | 674 HRESULT WASAPIAudioInputStream::InitializeAudioEngine() { |
602 DCHECK_EQ(OPEN_RESULT_OK, open_result_); | 675 DCHECK_EQ(OPEN_RESULT_OK, open_result_); |
603 DWORD flags; | 676 DWORD flags; |
604 // Use event-driven mode only fo regular input devices. For loopback the | 677 // Use event-driven mode only fo regular input devices. For loopback the |
605 // EVENTCALLBACK flag is specified when intializing | 678 // EVENTCALLBACK flag is specified when intializing |
606 // |audio_render_client_for_loopback_|. | 679 // |audio_render_client_for_loopback_|. |
607 if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId || | 680 if (device_id_ == AudioDeviceDescription::kLoopbackInputDeviceId || |
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731 | 804 |
732 return hr; | 805 return hr; |
733 } | 806 } |
734 | 807 |
735 void WASAPIAudioInputStream::ReportOpenResult() const { | 808 void WASAPIAudioInputStream::ReportOpenResult() const { |
736 DCHECK(!opened_); // This method must be called before we set this flag. | 809 DCHECK(!opened_); // This method must be called before we set this flag. |
737 UMA_HISTOGRAM_ENUMERATION("Media.Audio.Capture.Win.Open", open_result_, | 810 UMA_HISTOGRAM_ENUMERATION("Media.Audio.Capture.Win.Open", open_result_, |
738 OPEN_RESULT_MAX + 1); | 811 OPEN_RESULT_MAX + 1); |
739 } | 812 } |
740 | 813 |
814 double WASAPIAudioInputStream::ProvideInput(AudioBus* audio_bus, | |
815 uint32_t frames_delayed) { | |
816 convert_bus_->CopyTo(audio_bus); | |
817 data_was_converted_ = true; | |
818 return 1.0; | |
819 } | |
820 | |
741 } // namespace media | 821 } // namespace media |
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