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android / platform / external / chromium_org / 8bcbed8 / . / media / audio / win / audio_low_latency_output_win.cc
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// 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 "base/command_line.h"
#include "base/debug/trace_event.h"
#include "base/logging.h"
#include "base/memory/scoped_ptr.h"
#include "base/metrics/histogram.h"
#include "base/strings/utf_string_conversions.h"
#include "base/win/scoped_propvariant.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/limits.h"
#include "media/base/media_switches.h"
using base::win::ScopedComPtr;
using base::win::ScopedCOMInitializer;
using base::win::ScopedCoMem;
namespace media {
typedef uint32 ChannelConfig;
// Retrieves an integer mask which corresponds to the channel layout the
// audio engine uses for its internal processing/mixing of shared-mode
// streams. This mask indicates which channels are present in the multi-
// channel stream. The least significant bit corresponds with the Front Left
// speaker, the next least significant bit corresponds to the Front Right
// speaker, and so on, continuing in the order defined in KsMedia.h.
// See http://msdn.microsoft.com/en-us/library/windows/hardware/ff537083(v=vs.85).aspx
// for more details.
static ChannelConfig GetChannelConfig() {
WAVEFORMATPCMEX format;
return SUCCEEDED(CoreAudioUtil::GetDefaultSharedModeMixFormat(
eRender, eConsole, &format)) ?
static_cast<int>(format.dwChannelMask) : 0;
}
// Compare two sets of audio parameters and return true if they are equal.
// Note that bits_per_sample() is excluded from this comparison since Core
// Audio can deal with most bit depths. As an example, if the native/mixing
// bit depth is 32 bits (default), opening at 16 or 24 still works fine and
// the audio engine will do the required conversion for us. Channel count is
// excluded since Open() will fail anyways and it doesn't impact buffering.
static bool CompareAudioParametersNoBitDepthOrChannels(
const media::AudioParameters& a, const media::AudioParameters& b) {
return (a.format() == b.format() &&
a.sample_rate() == b.sample_rate() &&
a.frames_per_buffer() == b.frames_per_buffer());
}
// Converts Microsoft's channel configuration to ChannelLayout.
// This mapping is not perfect but the best we can do given the current
// ChannelLayout enumerator and the Windows-specific speaker configurations
// defined in ksmedia.h. Don't assume that the channel ordering in
// ChannelLayout is exactly the same as the Windows specific configuration.
// As an example: KSAUDIO_SPEAKER_7POINT1_SURROUND is mapped to
// CHANNEL_LAYOUT_7_1 but the positions of Back L, Back R and Side L, Side R
// speakers are different in these two definitions.
static ChannelLayout ChannelConfigToChannelLayout(ChannelConfig config) {
switch (config) {
case KSAUDIO_SPEAKER_DIRECTOUT:
return CHANNEL_LAYOUT_NONE;
case KSAUDIO_SPEAKER_MONO:
return CHANNEL_LAYOUT_MONO;
case KSAUDIO_SPEAKER_STEREO:
return CHANNEL_LAYOUT_STEREO;
case KSAUDIO_SPEAKER_QUAD:
return CHANNEL_LAYOUT_QUAD;
case KSAUDIO_SPEAKER_SURROUND:
return CHANNEL_LAYOUT_4_0;
case KSAUDIO_SPEAKER_5POINT1:
return CHANNEL_LAYOUT_5_1_BACK;
case KSAUDIO_SPEAKER_5POINT1_SURROUND:
return CHANNEL_LAYOUT_5_1;
case KSAUDIO_SPEAKER_7POINT1:
return CHANNEL_LAYOUT_7_1_WIDE;
case KSAUDIO_SPEAKER_7POINT1_SURROUND:
return CHANNEL_LAYOUT_7_1;
default:
VLOG(1) << "Unsupported channel layout: " << config;
return CHANNEL_LAYOUT_UNSUPPORTED;
}
}
// static
AUDCLNT_SHAREMODE WASAPIAudioOutputStream::GetShareMode() {
const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio))
return AUDCLNT_SHAREMODE_EXCLUSIVE;
return AUDCLNT_SHAREMODE_SHARED;
}
// static
int WASAPIAudioOutputStream::HardwareChannelCount() {
WAVEFORMATPCMEX format;
return SUCCEEDED(CoreAudioUtil::GetDefaultSharedModeMixFormat(
eRender, eConsole, &format)) ?
static_cast<int>(format.Format.nChannels) : 0;
}
// static
ChannelLayout WASAPIAudioOutputStream::HardwareChannelLayout() {
return ChannelConfigToChannelLayout(GetChannelConfig());
}
// static
int WASAPIAudioOutputStream::HardwareSampleRate(const std::string& device_id) {
WAVEFORMATPCMEX format;
ScopedComPtr<IAudioClient> client;
if (device_id.empty()) {
client = CoreAudioUtil::CreateDefaultClient(eRender, eConsole);
} else {
ScopedComPtr<IMMDevice> device(CoreAudioUtil::CreateDevice(device_id));
if (!device)
return 0;
client = CoreAudioUtil::CreateClient(device);
}
if (!client || FAILED(CoreAudioUtil::GetSharedModeMixFormat(client, &format)))
return 0;
return static_cast<int>(format.Format.nSamplesPerSec);
}
WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
const std::string& device_id,
const AudioParameters& params,
ERole device_role)
: creating_thread_id_(base::PlatformThread::CurrentId()),
manager_(manager),
opened_(false),
audio_parameters_are_valid_(false),
volume_(1.0),
endpoint_buffer_size_frames_(0),
device_id_(device_id),
device_role_(device_role),
share_mode_(GetShareMode()),
num_written_frames_(0),
source_(NULL),
audio_bus_(AudioBus::Create(params)) {
DCHECK(manager_);
VLOG(1) << "WASAPIAudioOutputStream::WASAPIAudioOutputStream()";
VLOG_IF(1, share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE)
<< "Core Audio (WASAPI) EXCLUSIVE MODE is enabled.";
if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
// Verify that the input audio parameters are identical (bit depth and
// channel count are excluded) to the preferred (native) audio parameters.
// Open() will fail if this is not the case.
AudioParameters preferred_params;
HRESULT hr = device_id_.empty() ?
CoreAudioUtil::GetPreferredAudioParameters(eRender, device_role,
&preferred_params) :
CoreAudioUtil::GetPreferredAudioParameters(device_id_,
&preferred_params);
audio_parameters_are_valid_ = SUCCEEDED(hr) &&
CompareAudioParametersNoBitDepthOrChannels(params, preferred_params);
LOG_IF(WARNING, !audio_parameters_are_valid_)
<< "Input and preferred parameters are not identical. "
<< "Device id: " << device_id_;
}
// 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";
// 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 = params.bits_per_sample();
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 = params.bits_per_sample();
format_.dwChannelMask = GetChannelConfig();
format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
// 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();
packet_size_ms_ = (1000.0 * packet_size_frames_) / params.sample_rate();
VLOG(1) << "Number of bytes per audio frame : " << format->nBlockAlign;
VLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
VLOG(1) << "Number of bytes per packet : " << packet_size_bytes_;
VLOG(1) << "Number of milliseconds per packet: " << packet_size_ms_;
// 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() {}
bool WASAPIAudioOutputStream::Open() {
VLOG(1) << "WASAPIAudioOutputStream::Open()";
DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
if (opened_)
return true;
// Audio parameters must be identical to the preferred set of parameters
// if shared mode (default) is utilized.
if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
if (!audio_parameters_are_valid_) {
LOG(ERROR) << "Audio parameters are not valid.";
return false;
}
}
// Create an IAudioClient interface for the default rendering IMMDevice.
ScopedComPtr<IAudioClient> audio_client;
if (device_id_.empty()) {
audio_client = CoreAudioUtil::CreateDefaultClient(eRender, device_role_);
} else {
ScopedComPtr<IMMDevice> device(CoreAudioUtil::CreateDevice(device_id_));
DLOG_IF(ERROR, !device) << "Failed to open device: " << device_id_;
if (device)
audio_client = CoreAudioUtil::CreateClient(device);
}
if (!audio_client)
return false;
// Extra sanity to ensure that the provided device format is still valid.
if (!CoreAudioUtil::IsFormatSupported(audio_client,
share_mode_,
&format_)) {
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, &format_, audio_samples_render_event_.Get(),
&endpoint_buffer_size_frames_);
if (FAILED(hr))
return false;
// We know from experience that the best possible callback sequence is
// achieved when the packet size (given by the native device period)
// is an even multiple of the endpoint buffer size.
// Examples: 48kHz => 960 % 480, 44.1kHz => 896 % 448 or 882 % 441.
if (endpoint_buffer_size_frames_ % packet_size_frames_ != 0) {
LOG(ERROR) << "Bailing out due to non-perfect timing.";
return false;
}
} else {
// TODO(henrika): break out to CoreAudioUtil::ExclusiveModeInitialize()
// when removing the enable-exclusive-audio flag.
hr = ExclusiveModeInitialization(audio_client,
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.
ScopedComPtr<IAudioRenderClient> audio_render_client =
CoreAudioUtil::CreateRenderClient(audio_client);
if (!audio_render_client)
return false;
// Store valid COM interfaces.
audio_client_ = audio_client;
audio_render_client_ = audio_render_client;
opened_ = true;
return true;
}
void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
VLOG(1) << "WASAPIAudioOutputStream::Start()";
DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
CHECK(callback);
CHECK(opened_);
if (render_thread_) {
CHECK_EQ(callback, source_);
return;
}
source_ = callback;
// 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"));
render_thread_->Start();
if (!render_thread_->HasBeenStarted()) {
LOG(ERROR) << "Failed to start WASAPI render thread.";
StopThread();
callback->OnError(this);
return;
}
// Ensure that the endpoint buffer is prepared with silence.
if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
if (!CoreAudioUtil::FillRenderEndpointBufferWithSilence(
audio_client_, audio_render_client_)) {
LOG(ERROR) << "Failed to prepare endpoint buffers with silence.";
StopThread();
callback->OnError(this);
return;
}
}
num_written_frames_ = endpoint_buffer_size_frames_;
// Start streaming data between the endpoint buffer and the audio engine.
HRESULT hr = audio_client_->Start();
if (FAILED(hr)) {
LOG_GETLASTERROR(ERROR)
<< "Failed to start output streaming: " << std::hex << hr;
StopThread();
callback->OnError(this);
}
}
void WASAPIAudioOutputStream::Stop() {
VLOG(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)) {
LOG_GETLASTERROR(ERROR)
<< "Failed to stop output streaming: " << std::hex << hr;
source_->OnError(this);
}
// 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)) {
LOG_GETLASTERROR(ERROR)
<< "Failed to reset streaming: " << std::hex << hr;
callback->OnError(this);
}
// 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() {
VLOG(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) {
VLOG(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) {
VLOG(1) << "GetVolume()";
*volume = static_cast<double>(volume_);
}
void WASAPIAudioOutputStream::Run() {
ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
// Increase the thread priority.
render_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
// 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_,
audio_samples_render_event_ };
UINT64 device_frequency = 0;
// The IAudioClock interface enables us to monitor a stream's data
// rate and the current position in the stream. Allocate it before we
// start spinning.
ScopedComPtr<IAudioClock> audio_clock;
hr = audio_client_->GetService(__uuidof(IAudioClock),
audio_clock.ReceiveVoid());
if (SUCCEEDED(hr)) {
// 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(arraysize(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(audio_clock, device_frequency);
break;
default:
error = true;
break;
}
}
if (playing && error) {
// 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();
PLOG(ERROR) << "WASAPI rendering failed.";
}
// Disable MMCSS.
if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
PLOG(WARNING) << "Failed to disable MMCSS";
}
}
bool WASAPIAudioOutputStream::RenderAudioFromSource(
IAudioClock* audio_clock, UINT64 device_frequency) {
TRACE_EVENT0("audio", "RenderAudioFromSource");
HRESULT hr = S_FALSE;
UINT32 num_queued_frames = 0;
uint8* 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 (num_available_frames < packet_size_frames_)
return true;
DLOG_IF(ERROR, num_available_frames % packet_size_frames_ != 0)
<< "Non-perfect timing detected (num_available_frames="
<< num_available_frames << ", packet_size_frames="
<< packet_size_frames_ << ")";
// Derive the number of packets we need to get from the client to
// fill up the available area in the endpoint buffer.
// |num_packets| will always be one for exclusive-mode streams and
// will be one in most cases for shared mode streams as well.
// However, we have found that two packets can sometimes be
// required.
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;
int audio_delay_bytes = 0;
hr = audio_clock->GetPosition(&position, NULL);
if (SUCCEEDED(hr)) {
// Stream position of the sample that is currently playing
// through the speaker.
double pos_sample_playing_frames = format_.Format.nSamplesPerSec *
(static_cast<double>(position) / device_frequency);
// Stream position of the last sample written to the endpoint
// buffer. Note that, the packet we are about to receive in
// the upcoming callback is also included.
size_t pos_last_sample_written_frames =
num_written_frames_ + packet_size_frames_;
// Derive the actual delay value which will be fed to the
// render client using the OnMoreData() callback.
audio_delay_bytes = (pos_last_sample_written_frames -
pos_sample_playing_frames) * format_.Format.nBlockAlign;
}
// Read a data packet from the registered client source and
// deliver a delay estimate in the same callback to the client.
// A time stamp is also stored in the AudioBuffersState. This
// time stamp can be used at the client side to compensate for
// the delay between the usage of the delay value and the time
// of generation.
int frames_filled = source_->OnMoreData(
audio_bus_.get(), AudioBuffersState(0, audio_delay_bytes));
uint32 num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
DCHECK_LE(num_filled_bytes, packet_size_bytes_);
// Note: If this ever changes to output raw float the data must be
// clipped and sanitized since it may come from an untrusted
// source such as NaCl.
const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
audio_bus_->Scale(volume_);
audio_bus_->ToInterleaved(
frames_filled, bytes_per_sample, 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* 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;
VLOG(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);
VLOG(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.
VLOG(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)) {
VLOG(1) << "IAudioClient::SetEventHandle: " << std::hex << hr;
return hr;
}
}
UINT32 buffer_size_in_frames = 0;
hr = client->GetBufferSize(&buffer_size_in_frames);
if (FAILED(hr)) {
VLOG(1) << "IAudioClient::GetBufferSize: " << std::hex << hr;
return hr;
}
*endpoint_buffer_size = buffer_size_in_frames;
VLOG(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