| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "media/audio/win/audio_low_latency_output_win.h" |
| |
| #include <Functiondiscoverykeys_devpkey.h> |
| #include <objbase.h> |
| |
| #include <climits> |
| |
| #include "base/command_line.h" |
| #include "base/logging.h" |
| #include "base/metrics/histogram.h" |
| #include "base/stl_util.h" |
| #include "base/strings/utf_string_conversions.h" |
| #include "base/time/time.h" |
| #include "base/trace_event/trace_event.h" |
| #include "base/win/scoped_propvariant.h" |
| #include "media/audio/audio_device_description.h" |
| #include "media/audio/win/audio_manager_win.h" |
| #include "media/audio/win/avrt_wrapper_win.h" |
| #include "media/audio/win/core_audio_util_win.h" |
| #include "media/base/audio_sample_types.h" |
| #include "media/base/limits.h" |
| #include "media/base/media_switches.h" |
| |
| using base::win::ScopedCOMInitializer; |
| using base::win::ScopedCoMem; |
| |
| namespace media { |
| |
| // static |
| AUDCLNT_SHAREMODE WASAPIAudioOutputStream::GetShareMode() { |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) |
| return AUDCLNT_SHAREMODE_EXCLUSIVE; |
| return AUDCLNT_SHAREMODE_SHARED; |
| } |
| |
| WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager, |
| const std::string& device_id, |
| const AudioParameters& params, |
| ERole device_role) |
| : creating_thread_id_(base::PlatformThread::CurrentId()), |
| manager_(manager), |
| format_(), |
| opened_(false), |
| volume_(1.0), |
| packet_size_frames_(0), |
| requested_iaudioclient3_buffer_size_(0), |
| packet_size_bytes_(0), |
| endpoint_buffer_size_frames_(0), |
| device_id_(device_id), |
| device_role_(device_role), |
| share_mode_(GetShareMode()), |
| num_written_frames_(0), |
| source_(NULL) { |
| DCHECK(manager_); |
| |
| // The empty string is used to indicate a default device and the |
| // |device_role_| member controls whether that's the default or default |
| // communications device. |
| DCHECK_NE(device_id_, AudioDeviceDescription::kDefaultDeviceId); |
| DCHECK_NE(device_id_, AudioDeviceDescription::kCommunicationsDeviceId); |
| |
| DVLOG(1) << "WASAPIAudioOutputStream::WASAPIAudioOutputStream()"; |
| DVLOG_IF(1, share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE) |
| << "Core Audio (WASAPI) EXCLUSIVE MODE is enabled."; |
| |
| // Load the Avrt DLL if not already loaded. Required to support MMCSS. |
| bool avrt_init = avrt::Initialize(); |
| DCHECK(avrt_init) << "Failed to load the avrt.dll"; |
| |
| audio_bus_ = AudioBus::Create(params); |
| |
| // Set up the desired render format specified by the client. We use the |
| // WAVE_FORMAT_EXTENSIBLE structure to ensure that multiple channel ordering |
| // and high precision data can be supported. |
| |
| // Begin with the WAVEFORMATEX structure that specifies the basic format. |
| WAVEFORMATEX* format = &format_.Format; |
| format->wFormatTag = WAVE_FORMAT_EXTENSIBLE; |
| format->nChannels = params.channels(); |
| format->nSamplesPerSec = params.sample_rate(); |
| format->wBitsPerSample = sizeof(float) * 8; |
| format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels; |
| format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign; |
| format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX); |
| |
| // Add the parts which are unique to WAVE_FORMAT_EXTENSIBLE. |
| format_.Samples.wValidBitsPerSample = format->wBitsPerSample; |
| format_.dwChannelMask = CoreAudioUtil::GetChannelConfig(device_id, eRender); |
| format_.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; |
| |
| // 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); |
| DVLOG(1) << "Number of bytes per audio frame : " << format->nBlockAlign; |
| DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_; |
| DVLOG(1) << "Number of bytes per packet : " << packet_size_bytes_; |
| DVLOG(1) << "Number of milliseconds per packet: " |
| << params.GetBufferDuration().InMillisecondsF(); |
| |
| AudioParameters::HardwareCapabilities hardware_capabilities = |
| params.hardware_capabilities().value_or( |
| AudioParameters::HardwareCapabilities()); |
| |
| // Only request an explicit buffer size if we are requesting the minimum |
| // supported by the hardware, everything else uses the older IAudioClient API. |
| if (params.frames_per_buffer() == |
| hardware_capabilities.min_frames_per_buffer) { |
| requested_iaudioclient3_buffer_size_ = |
| hardware_capabilities.min_frames_per_buffer; |
| } |
| |
| // All events are auto-reset events and non-signaled initially. |
| |
| // Create the event which the audio engine will signal each time |
| // a buffer becomes ready to be processed by the client. |
| audio_samples_render_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); |
| DCHECK(audio_samples_render_event_.IsValid()); |
| |
| // Create the event which will be set in Stop() when capturing shall stop. |
| stop_render_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL)); |
| DCHECK(stop_render_event_.IsValid()); |
| } |
| |
| WASAPIAudioOutputStream::~WASAPIAudioOutputStream() { |
| DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_); |
| } |
| |
| bool WASAPIAudioOutputStream::Open() { |
| DVLOG(1) << "WASAPIAudioOutputStream::Open()"; |
| DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_); |
| if (opened_) |
| return true; |
| |
| DCHECK(!audio_client_.Get()); |
| DCHECK(!audio_render_client_.Get()); |
| |
| const bool communications_device = |
| device_id_.empty() ? (device_role_ == eCommunications) : false; |
| |
| Microsoft::WRL::ComPtr<IAudioClient> audio_client( |
| CoreAudioUtil::CreateClient(device_id_, eRender, device_role_)); |
| |
| if (!audio_client.Get()) |
| return false; |
| |
| // Extra sanity to ensure that the provided device format is still valid. |
| if (!CoreAudioUtil::IsFormatSupported(audio_client.Get(), share_mode_, |
| &format_)) { |
| LOG(ERROR) << "Audio parameters are not supported."; |
| return false; |
| } |
| |
| HRESULT hr = S_FALSE; |
| if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) { |
| // Initialize the audio stream between the client and the device in shared |
| // mode and using event-driven buffer handling. |
| hr = CoreAudioUtil::SharedModeInitialize( |
| audio_client.Get(), &format_, audio_samples_render_event_.Get(), |
| requested_iaudioclient3_buffer_size_, &endpoint_buffer_size_frames_, |
| communications_device ? &kCommunicationsSessionId : NULL); |
| if (FAILED(hr)) |
| return false; |
| |
| REFERENCE_TIME device_period = 0; |
| if (FAILED(CoreAudioUtil::GetDevicePeriod( |
| audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &device_period))) { |
| return false; |
| } |
| |
| const int preferred_frames_per_buffer = static_cast<int>( |
| format_.Format.nSamplesPerSec * |
| CoreAudioUtil::ReferenceTimeToTimeDelta(device_period) |
| .InSecondsF() + |
| 0.5); |
| |
| // Packet size should always be an even divisor of the device period for |
| // best performance; things will still work otherwise, but may glitch for a |
| // couple of reasons. |
| // |
| // The first reason is if/when repeated RenderAudioFromSource() hit the |
| // shared memory boundary between the renderer and the browser. The next |
| // audio buffer is always requested after the current request is consumed. |
| // With back-to-back calls the round-trip may not be fast enough and thus |
| // audio will glitch as we fail to deliver audio in a timely manner. |
| // |
| // The second reason is event wakeup efficiency. We may have too few or too |
| // many frames to fill the output buffer requested by WASAPI. If too few, |
| // we'll refuse the render event and wait until more output space is |
| // available. If we have too many frames, we'll only partially fill and |
| // wait for the next render event. In either case certain remainders may |
| // leave us unable to fulfill the request in a timely manner, thus glitches. |
| // |
| // Log a warning in these cases so we can help users in the field. |
| // Examples: 48kHz => 960 % 480, 44.1kHz => 896 % 448 or 882 % 441. |
| if (preferred_frames_per_buffer % packet_size_frames_) { |
| LOG(WARNING) |
| << "Using WASAPI output with a non-optimal buffer size, glitches from" |
| << " back to back shared memory reads and partial fills of WASAPI" |
| << " output buffers may occur. Buffer size of " |
| << packet_size_frames_ << " is not an even divisor of " |
| << preferred_frames_per_buffer; |
| } |
| } else { |
| // TODO(henrika): break out to CoreAudioUtil::ExclusiveModeInitialize() |
| // when removing the enable-exclusive-audio flag. |
| hr = ExclusiveModeInitialization(audio_client.Get(), |
| audio_samples_render_event_.Get(), |
| &endpoint_buffer_size_frames_); |
| if (FAILED(hr)) |
| return false; |
| |
| // The buffer scheme for exclusive mode streams is not designed for max |
| // flexibility. We only allow a "perfect match" between the packet size set |
| // by the user and the actual endpoint buffer size. |
| if (endpoint_buffer_size_frames_ != packet_size_frames_) { |
| LOG(ERROR) << "Bailing out due to non-perfect timing."; |
| return false; |
| } |
| } |
| |
| // Create an IAudioRenderClient client for an initialized IAudioClient. |
| // The IAudioRenderClient interface enables us to write output data to |
| // a rendering endpoint buffer. |
| Microsoft::WRL::ComPtr<IAudioRenderClient> audio_render_client = |
| CoreAudioUtil::CreateRenderClient(audio_client.Get()); |
| if (!audio_render_client.Get()) |
| return false; |
| |
| // Store valid COM interfaces. |
| audio_client_ = audio_client; |
| audio_render_client_ = audio_render_client; |
| |
| hr = audio_client_->GetService(IID_PPV_ARGS(&audio_clock_)); |
| if (FAILED(hr)) { |
| LOG(ERROR) << "Failed to get IAudioClock service."; |
| return false; |
| } |
| |
| opened_ = true; |
| return true; |
| } |
| |
| void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) { |
| DVLOG(1) << "WASAPIAudioOutputStream::Start()"; |
| DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_); |
| CHECK(callback); |
| CHECK(opened_); |
| |
| if (render_thread_) { |
| CHECK_EQ(callback, source_); |
| return; |
| } |
| |
| // Ensure that the endpoint buffer is prepared with silence. Also serves as |
| // a sanity check for the IAudioClient and IAudioRenderClient which may have |
| // been invalidated by Windows since the last Stop() call. |
| // |
| // While technically we only need to retry when WASAPI tells us the device has |
| // been invalidated (AUDCLNT_E_DEVICE_INVALIDATED), we retry for all errors |
| // for simplicity and due to large sites like YouTube reporting high success |
| // rates with a simple retry upon detection of an audio output error. |
| if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) { |
| if (!CoreAudioUtil::FillRenderEndpointBufferWithSilence( |
| audio_client_.Get(), audio_render_client_.Get())) { |
| DLOG(WARNING) << "Failed to prepare endpoint buffers with silence. " |
| "Attempting recovery with a new IAudioClient and " |
| "IAudioRenderClient."; |
| |
| opened_ = false; |
| audio_client_.Reset(); |
| audio_render_client_.Reset(); |
| if (!Open() || !CoreAudioUtil::FillRenderEndpointBufferWithSilence( |
| audio_client_.Get(), audio_render_client_.Get())) { |
| DLOG(ERROR) << "Failed recovery of audio clients; Start() failed."; |
| callback->OnError(); |
| return; |
| } |
| } |
| } |
| |
| source_ = callback; |
| num_written_frames_ = endpoint_buffer_size_frames_; |
| |
| // Create and start the thread that will drive the rendering by waiting for |
| // render events. |
| render_thread_.reset(new base::DelegateSimpleThread( |
| this, "wasapi_render_thread", |
| base::SimpleThread::Options(base::ThreadPriority::REALTIME_AUDIO))); |
| render_thread_->Start(); |
| if (!render_thread_->HasBeenStarted()) { |
| LOG(ERROR) << "Failed to start WASAPI render thread."; |
| StopThread(); |
| callback->OnError(); |
| return; |
| } |
| |
| // Start streaming data between the endpoint buffer and the audio engine. |
| HRESULT hr = audio_client_->Start(); |
| if (FAILED(hr)) { |
| PLOG(ERROR) << "Failed to start output streaming: " << std::hex << hr; |
| StopThread(); |
| callback->OnError(); |
| } |
| } |
| |
| void WASAPIAudioOutputStream::Stop() { |
| DVLOG(1) << "WASAPIAudioOutputStream::Stop()"; |
| DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_); |
| if (!render_thread_) |
| return; |
| |
| // Stop output audio streaming. |
| HRESULT hr = audio_client_->Stop(); |
| if (FAILED(hr)) { |
| PLOG(ERROR) << "Failed to stop output streaming: " << std::hex << hr; |
| source_->OnError(); |
| } |
| |
| // Make a local copy of |source_| since StopThread() will clear it. |
| AudioSourceCallback* callback = source_; |
| StopThread(); |
| |
| // Flush all pending data and reset the audio clock stream position to 0. |
| hr = audio_client_->Reset(); |
| if (FAILED(hr)) { |
| PLOG(ERROR) << "Failed to reset streaming: " << std::hex << hr; |
| callback->OnError(); |
| } |
| |
| // Extra safety check to ensure that the buffers are cleared. |
| // If the buffers are not cleared correctly, the next call to Start() |
| // would fail with AUDCLNT_E_BUFFER_ERROR at IAudioRenderClient::GetBuffer(). |
| // This check is is only needed for shared-mode streams. |
| if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) { |
| UINT32 num_queued_frames = 0; |
| audio_client_->GetCurrentPadding(&num_queued_frames); |
| DCHECK_EQ(0u, num_queued_frames); |
| } |
| } |
| |
| void WASAPIAudioOutputStream::Close() { |
| DVLOG(1) << "WASAPIAudioOutputStream::Close()"; |
| DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_); |
| |
| // It is valid to call Close() before calling open or Start(). |
| // It is also valid to call Close() after Start() has been called. |
| Stop(); |
| |
| // Inform the audio manager that we have been closed. This will cause our |
| // destruction. |
| manager_->ReleaseOutputStream(this); |
| } |
| |
| void WASAPIAudioOutputStream::SetVolume(double volume) { |
| DVLOG(1) << "SetVolume(volume=" << volume << ")"; |
| float volume_float = static_cast<float>(volume); |
| if (volume_float < 0.0f || volume_float > 1.0f) { |
| return; |
| } |
| volume_ = volume_float; |
| } |
| |
| void WASAPIAudioOutputStream::GetVolume(double* volume) { |
| DVLOG(1) << "GetVolume()"; |
| *volume = static_cast<double>(volume_); |
| } |
| |
| void WASAPIAudioOutputStream::Run() { |
| ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); |
| |
| // Enable MMCSS to ensure that this thread receives prioritized access to |
| // CPU resources. |
| DWORD task_index = 0; |
| HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio", |
| &task_index); |
| bool mmcss_is_ok = |
| (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL)); |
| if (!mmcss_is_ok) { |
| // Failed to enable MMCSS on this thread. It is not fatal but can lead |
| // to reduced QoS at high load. |
| DWORD err = GetLastError(); |
| LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ")."; |
| } |
| |
| HRESULT hr = S_FALSE; |
| |
| bool playing = true; |
| bool error = false; |
| HANDLE wait_array[] = { stop_render_event_.Get(), |
| audio_samples_render_event_.Get() }; |
| UINT64 device_frequency = 0; |
| |
| // The device frequency is the frequency generated by the hardware clock in |
| // the audio device. The GetFrequency() method reports a constant frequency. |
| hr = audio_clock_->GetFrequency(&device_frequency); |
| error = FAILED(hr); |
| PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: " |
| << std::hex << hr; |
| |
| // Keep rendering audio until the stop event or the stream-switch event |
| // is signaled. An error event can also break the main thread loop. |
| while (playing && !error) { |
| // Wait for a close-down event, stream-switch event or a new render event. |
| DWORD wait_result = WaitForMultipleObjects(base::size(wait_array), |
| wait_array, FALSE, INFINITE); |
| |
| switch (wait_result) { |
| case WAIT_OBJECT_0 + 0: |
| // |stop_render_event_| has been set. |
| playing = false; |
| break; |
| case WAIT_OBJECT_0 + 1: |
| // |audio_samples_render_event_| has been set. |
| error = !RenderAudioFromSource(device_frequency); |
| break; |
| default: |
| error = true; |
| break; |
| } |
| } |
| |
| if (playing && error) { |
| LOG(ERROR) << "WASAPI rendering failed."; |
| |
| // Stop audio rendering since something has gone wrong in our main thread |
| // loop. Note that, we are still in a "started" state, hence a Stop() call |
| // is required to join the thread properly. |
| audio_client_->Stop(); |
| |
| // Notify clients that something has gone wrong and that this stream should |
| // be destroyed instead of reused in the future. |
| source_->OnError(); |
| } |
| |
| // Disable MMCSS. |
| if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) { |
| PLOG(WARNING) << "Failed to disable MMCSS"; |
| } |
| } |
| |
| bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) { |
| TRACE_EVENT0("audio", "RenderAudioFromSource"); |
| |
| HRESULT hr = S_FALSE; |
| UINT32 num_queued_frames = 0; |
| uint8_t* audio_data = NULL; |
| |
| // Contains how much new data we can write to the buffer without |
| // the risk of overwriting previously written data that the audio |
| // engine has not yet read from the buffer. |
| size_t num_available_frames = 0; |
| |
| if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) { |
| // Get the padding value which represents the amount of rendering |
| // data that is queued up to play in the endpoint buffer. |
| hr = audio_client_->GetCurrentPadding(&num_queued_frames); |
| num_available_frames = |
| endpoint_buffer_size_frames_ - num_queued_frames; |
| if (FAILED(hr)) { |
| DLOG(ERROR) << "Failed to retrieve amount of available space: " |
| << std::hex << hr; |
| return false; |
| } |
| } else { |
| // While the stream is running, the system alternately sends one |
| // buffer or the other to the client. This form of double buffering |
| // is referred to as "ping-ponging". Each time the client receives |
| // a buffer from the system (triggers this event) the client must |
| // process the entire buffer. Calls to the GetCurrentPadding method |
| // are unnecessary because the packet size must always equal the |
| // buffer size. In contrast to the shared mode buffering scheme, |
| // the latency for an event-driven, exclusive-mode stream depends |
| // directly on the buffer size. |
| num_available_frames = endpoint_buffer_size_frames_; |
| } |
| |
| // Check if there is enough available space to fit the packet size |
| // specified by the client. If not, wait until a future callback. |
| if (num_available_frames < packet_size_frames_) |
| return true; |
| |
| // Derive the number of packets we need to get from the client to fill up the |
| // available area in the endpoint buffer. Well-behaved (> Vista) clients and |
| // exclusive mode streams should generally have a |num_packets| value of 1. |
| // |
| // Vista clients are not able to maintain reliable callbacks, so the endpoint |
| // buffer may exhaust itself such that back-to-back callbacks are occasionally |
| // necessary to avoid glitches. In such cases we have no choice but to issue |
| // back-to-back reads and pray that the browser side has enough data cached or |
| // that the render can fulfill the read before we glitch anyways. |
| // |
| // API documentation does not guarantee that even on Win7+ clients we won't |
| // need to fill more than a period size worth of buffers; but in practice this |
| // appears to be infrequent. |
| // |
| // See http://crbug.com/524947. |
| const size_t num_packets = num_available_frames / packet_size_frames_; |
| for (size_t n = 0; n < num_packets; ++n) { |
| // Grab all available space in the rendering endpoint buffer |
| // into which the client can write a data packet. |
| hr = audio_render_client_->GetBuffer(packet_size_frames_, |
| &audio_data); |
| if (FAILED(hr)) { |
| DLOG(ERROR) << "Failed to use rendering audio buffer: " |
| << std::hex << hr; |
| return false; |
| } |
| |
| // Derive the audio delay which corresponds to the delay between |
| // a render event and the time when the first audio sample in a |
| // packet is played out through the speaker. This delay value |
| // can typically be utilized by an acoustic echo-control (AEC) |
| // unit at the render side. |
| UINT64 position = 0; |
| UINT64 qpc_position = 0; |
| base::TimeDelta delay; |
| base::TimeTicks delay_timestamp; |
| hr = audio_clock_->GetPosition(&position, &qpc_position); |
| if (SUCCEEDED(hr)) { |
| // Number of frames already played out through the speaker. |
| const uint64_t played_out_frames = |
| format_.Format.nSamplesPerSec * position / device_frequency; |
| |
| // Number of frames that have been written to the buffer but not yet |
| // played out. |
| const uint64_t delay_frames = num_written_frames_ - played_out_frames; |
| |
| // Convert the delay from frames to time. |
| delay = base::TimeDelta::FromMicroseconds( |
| delay_frames * base::Time::kMicrosecondsPerSecond / |
| format_.Format.nSamplesPerSec); |
| // Note: the obtained |qpc_position| value is in 100ns intervals and from |
| // the same time origin as QPC. We can simply convert it into us dividing |
| // by 10.0 since 10x100ns = 1us. |
| delay_timestamp += base::TimeDelta::FromMicroseconds(qpc_position * 0.1); |
| } else { |
| // Use a delay of zero. |
| delay_timestamp = base::TimeTicks::Now(); |
| } |
| |
| // Read a data packet from the registered client source and |
| // deliver a delay estimate in the same callback to the client. |
| |
| 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_); |
| audio_bus_->ToInterleaved<Float32SampleTypeTraits>( |
| frames_filled, reinterpret_cast<float*>(audio_data)); |
| |
| // Release the buffer space acquired in the GetBuffer() call. |
| // Render silence if we were not able to fill up the buffer totally. |
| DWORD flags = (num_filled_bytes < packet_size_bytes_) ? |
| AUDCLNT_BUFFERFLAGS_SILENT : 0; |
| audio_render_client_->ReleaseBuffer(packet_size_frames_, flags); |
| |
| num_written_frames_ += packet_size_frames_; |
| } |
| |
| return true; |
| } |
| |
| HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization( |
| IAudioClient* client, |
| HANDLE event_handle, |
| uint32_t* endpoint_buffer_size) { |
| DCHECK_EQ(share_mode_, AUDCLNT_SHAREMODE_EXCLUSIVE); |
| |
| float f = (1000.0 * packet_size_frames_) / format_.Format.nSamplesPerSec; |
| REFERENCE_TIME requested_buffer_duration = |
| static_cast<REFERENCE_TIME>(f * 10000.0 + 0.5); |
| |
| DWORD stream_flags = AUDCLNT_STREAMFLAGS_NOPERSIST; |
| bool use_event = (event_handle != NULL && |
| event_handle != INVALID_HANDLE_VALUE); |
| if (use_event) |
| stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK; |
| DVLOG(2) << "stream_flags: 0x" << std::hex << stream_flags; |
| |
| // Initialize the audio stream between the client and the device. |
| // For an exclusive-mode stream that uses event-driven buffering, the |
| // caller must specify nonzero values for hnsPeriodicity and |
| // hnsBufferDuration, and the values of these two parameters must be equal. |
| // The Initialize method allocates two buffers for the stream. Each buffer |
| // is equal in duration to the value of the hnsBufferDuration parameter. |
| // Following the Initialize call for a rendering stream, the caller should |
| // fill the first of the two buffers before starting the stream. |
| HRESULT hr = S_FALSE; |
| hr = client->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE, |
| stream_flags, |
| requested_buffer_duration, |
| requested_buffer_duration, |
| reinterpret_cast<WAVEFORMATEX*>(&format_), |
| NULL); |
| if (FAILED(hr)) { |
| if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) { |
| LOG(ERROR) << "AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED"; |
| |
| UINT32 aligned_buffer_size = 0; |
| client->GetBufferSize(&aligned_buffer_size); |
| DVLOG(1) << "Use aligned buffer size instead: " << aligned_buffer_size; |
| |
| // Calculate new aligned periodicity. Each unit of reference time |
| // is 100 nanoseconds. |
| REFERENCE_TIME aligned_buffer_duration = static_cast<REFERENCE_TIME>( |
| (10000000.0 * aligned_buffer_size / format_.Format.nSamplesPerSec) |
| + 0.5); |
| |
| // It is possible to re-activate and re-initialize the audio client |
| // at this stage but we bail out with an error code instead and |
| // combine it with a log message which informs about the suggested |
| // aligned buffer size which should be used instead. |
| DVLOG(1) << "aligned_buffer_duration: " |
| << static_cast<double>(aligned_buffer_duration / 10000.0) |
| << " [ms]"; |
| } else if (hr == AUDCLNT_E_INVALID_DEVICE_PERIOD) { |
| // We will get this error if we try to use a smaller buffer size than |
| // the minimum supported size (usually ~3ms on Windows 7). |
| LOG(ERROR) << "AUDCLNT_E_INVALID_DEVICE_PERIOD"; |
| } |
| return hr; |
| } |
| |
| if (use_event) { |
| hr = client->SetEventHandle(event_handle); |
| if (FAILED(hr)) { |
| DVLOG(1) << "IAudioClient::SetEventHandle: " << std::hex << hr; |
| return hr; |
| } |
| } |
| |
| UINT32 buffer_size_in_frames = 0; |
| hr = client->GetBufferSize(&buffer_size_in_frames); |
| if (FAILED(hr)) { |
| DVLOG(1) << "IAudioClient::GetBufferSize: " << std::hex << hr; |
| return hr; |
| } |
| |
| *endpoint_buffer_size = buffer_size_in_frames; |
| DVLOG(2) << "endpoint buffer size: " << buffer_size_in_frames; |
| return hr; |
| } |
| |
| void WASAPIAudioOutputStream::StopThread() { |
| if (render_thread_) { |
| if (render_thread_->HasBeenStarted()) { |
| // Wait until the thread completes and perform cleanup. |
| SetEvent(stop_render_event_.Get()); |
| render_thread_->Join(); |
| } |
| |
| render_thread_.reset(); |
| |
| // Ensure that we don't quit the main thread loop immediately next |
| // time Start() is called. |
| ResetEvent(stop_render_event_.Get()); |
| } |
| |
| source_ = NULL; |
| } |
| |
| } // namespace media |