Chromium Code Reviews| Index: remoting/host/audio_capturer_win.cc |
| diff --git a/remoting/host/audio_capturer_win.cc b/remoting/host/audio_capturer_win.cc |
| index 2610e6e2b692eed2523b0dfe2047d3222b379996..361352a6cc3bf275b3216c294eabbe2af47934ea 100644 |
| --- a/remoting/host/audio_capturer_win.cc |
| +++ b/remoting/host/audio_capturer_win.cc |
| @@ -21,7 +21,6 @@ |
| #include "remoting/host/win/default_audio_device_change_detector.h" |
| namespace { |
| -const int kChannels = 2; |
| const int kBytesPerSample = 2; |
| const int kBitsPerSample = kBytesPerSample * 8; |
| // Conversion factor from 100ns to 1ms. |
| @@ -39,6 +38,21 @@ const int kMinTimerInterval = 30; |
| // Upper bound for the timer precision error, in milliseconds. |
| // Timers are supposed to be accurate to 20ms, so we use 30ms to be safe. |
| const int kMaxExpectedTimerLag = 30; |
| + |
| +// Chooses a similar and supported channel configuration. |channels| represents |
| +// the selected channels. It can be nullptr. |
| +void ChooseChannels(WORD* channel_count, DWORD* channels) { |
| + // We supports mono to 7.1. |
| + if (*channel_count >= 1 && *channel_count <= 8) { |
| + return; |
| + } |
| + |
| + *channel_count = remoting::AudioPacket::CHANNELS_STEREO; |
| + if (channels) { |
| + *channels = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; |
| + } |
| +} |
| + |
| } // namespace |
| namespace remoting { |
| @@ -163,11 +177,12 @@ bool AudioCapturerWin::Initialize() { |
| wave_format_ex_->nSamplesPerSec); |
| wave_format_ex_->wFormatTag = WAVE_FORMAT_PCM; |
| - wave_format_ex_->nChannels = kChannels; |
| + ChooseChannels(&wave_format_ex_->nChannels, nullptr); |
| wave_format_ex_->wBitsPerSample = kBitsPerSample; |
| - wave_format_ex_->nBlockAlign = kChannels * kBytesPerSample; |
| + wave_format_ex_->nBlockAlign = |
| + wave_format_ex_->nChannels * kBytesPerSample; |
| wave_format_ex_->nAvgBytesPerSec = |
| - sampling_rate_ * kChannels * kBytesPerSample; |
| + sampling_rate_ * wave_format_ex_->nChannels * kBytesPerSample; |
| break; |
| case WAVE_FORMAT_EXTENSIBLE: { |
| PWAVEFORMATEXTENSIBLE wave_format_extensible = |
| @@ -186,13 +201,15 @@ bool AudioCapturerWin::Initialize() { |
| wave_format_extensible->SubFormat = KSDATAFORMAT_SUBTYPE_PCM; |
| wave_format_extensible->Samples.wValidBitsPerSample = kBitsPerSample; |
| - wave_format_extensible->Format.nChannels = kChannels; |
| + ChooseChannels(&wave_format_extensible->Format.nChannels, |
| + &wave_format_extensible->dwChannelMask); |
| wave_format_extensible->Format.nSamplesPerSec = sampling_rate_; |
| wave_format_extensible->Format.wBitsPerSample = kBitsPerSample; |
| wave_format_extensible->Format.nBlockAlign = |
| - kChannels * kBytesPerSample; |
| + wave_format_extensible->Format.nChannels * kBytesPerSample; |
| wave_format_extensible->Format.nAvgBytesPerSec = |
| - sampling_rate_ * kChannels * kBytesPerSample; |
| + sampling_rate_ * wave_format_extensible->Format.nChannels * |
| + kBytesPerSample; |
| } else { |
| LOG(ERROR) << "Failed to force 16-bit samples"; |
| return false; |
| @@ -233,7 +250,7 @@ bool AudioCapturerWin::Initialize() { |
| } |
| volume_filter_.ActivateBy(mm_device_.Get()); |
| - volume_filter_.Initialize(sampling_rate_, kChannels); |
| + volume_filter_.Initialize(sampling_rate_, wave_format_ex_->nChannels); |
| return true; |
| } |
| @@ -280,7 +297,8 @@ void AudioCapturerWin::DoCapture() { |
| packet->set_encoding(AudioPacket::ENCODING_RAW); |
| packet->set_sampling_rate(sampling_rate_); |
| packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2); |
| - packet->set_channels(AudioPacket::CHANNELS_STEREO); |
| + packet->set_channels(static_cast<AudioPacket::Channels>( |
|
Sergey Ulanov
2017/06/05 18:03:19
I think this would be the right place to add a com
Hzj_jie
2017/06/06 03:21:43
Done.
|
| + wave_format_ex_->nChannels)); |
| callback_.Run(std::move(packet)); |
| } |