diff options
Diffstat (limited to 'chromium/media/audio/win/audio_low_latency_output_win.cc')
| -rw-r--r-- | chromium/media/audio/win/audio_low_latency_output_win.cc | 50 |
1 files changed, 46 insertions, 4 deletions
diff --git a/chromium/media/audio/win/audio_low_latency_output_win.cc b/chromium/media/audio/win/audio_low_latency_output_win.cc index c3a4facf84d..5767a6bafa1 100644 --- a/chromium/media/audio/win/audio_low_latency_output_win.cc +++ b/chromium/media/audio/win/audio_low_latency_output_win.cc @@ -1,4 +1,4 @@ -// Copyright (c) 2012 The Chromium Authors. All rights reserved. +// Copyright 2012 The Chromium Authors // Use of this source code is governed by a BSD-style license that can be // found in the LICENSE file. @@ -89,6 +89,7 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream( manager_(manager), glitch_reporter_(SystemGlitchReporter::StreamType::kRender), format_(), + params_(params), opened_(false), volume_(1.0), packet_size_frames_(0), @@ -102,6 +103,10 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream( source_(nullptr), log_callback_(std::move(log_callback)) { DCHECK(manager_); +#if BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) + if (params.format() == AudioParameters::AUDIO_BITSTREAM_DTS) + DCHECK_EQ(GetShareMode(), AUDCLNT_SHAREMODE_EXCLUSIVE); +#endif // BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) // The empty string is used to indicate a default device and the // |device_role_| member controls whether that's the default or default @@ -138,13 +143,28 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream( format_.Samples.wValidBitsPerSample = format->wBitsPerSample; format_.dwChannelMask = CoreAudioUtil::GetChannelConfig(device_id, eRender); format_.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; - SendLogMessage("%s => (audio engine format=[%s])", __func__, - CoreAudioUtil::WaveFormatToString(&format_).c_str()); // Store size (in different units) of audio packets which we expect to // get from the audio endpoint device in each render event. packet_size_frames_ = params.frames_per_buffer(); packet_size_bytes_ = params.GetBytesPerBuffer(kSampleFormatF32); + +#if BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) + if (params.format() == AudioParameters::AUDIO_BITSTREAM_DTS) { + format_.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DTS; + format->wBitsPerSample = 16; + format->nChannels = 2; + format_.dwChannelMask = KSAUDIO_SPEAKER_STEREO; + format_.Samples.wValidBitsPerSample = format->wBitsPerSample; + format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels; + format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign; + packet_size_frames_ = 512; + packet_size_bytes_ = params.GetBytesPerBuffer(kSampleFormatS16); + } +#endif // BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) + SendLogMessage("%s => (audio engine format=[%s])", __func__, + CoreAudioUtil::WaveFormatToString(&format_).c_str()); + SendLogMessage("%s => (packet size=[%zu bytes/%zu audio frames/%.3f ms])", __func__, packet_size_bytes_, packet_size_frames_, params.GetBufferDuration().InMillisecondsF()); @@ -462,6 +482,10 @@ void WASAPIAudioOutputStream::Close() { void WASAPIAudioOutputStream::Flush() {} void WASAPIAudioOutputStream::SetVolume(double volume) { +#if BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) + if (params_.format() == AudioParameters::AUDIO_BITSTREAM_DTS) + return; +#endif // BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) SendLogMessage("%s({volume=%.2f})", __func__, volume); float volume_float = static_cast<float>(volume); if (volume_float < 0.0f || volume_float > 1.0f) { @@ -707,11 +731,29 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) { // Read a data packet from the registered client source and // deliver a delay estimate in the same callback to the client. +#if BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) + if (params_.format() == AudioParameters::AUDIO_BITSTREAM_DTS) { + std::unique_ptr<AudioBus> audio_bus( + AudioBus::WrapMemory(params_, audio_data)); + audio_bus_->set_is_bitstream_format(true); + int frames_filled = + source_->OnMoreData(delay, delay_timestamp, 0, audio_bus.get()); + + // During pause/seek, keep the pipeline filled with zero'ed frames. + if (!frames_filled) + memset(audio_data, 0, packet_size_frames_); + + // Release the buffer space acquired in the GetBuffer() call. + // Render silence if we were not able to fill up the buffer totally. + audio_render_client_->ReleaseBuffer(packet_size_frames_, 0); + num_written_frames_ += packet_size_frames_; + return true; + } +#endif // BUILDFLAG(ENABLE_PLATFORM_DTS_AUDIO) int frames_filled = source_->OnMoreData(delay, delay_timestamp, 0, audio_bus_.get()); uint32_t num_filled_bytes = frames_filled * format_.Format.nBlockAlign; DCHECK_LE(num_filled_bytes, packet_size_bytes_); - audio_bus_->Scale(volume_); // We skip clipping since that occurs at the shared memory boundary. |