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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_input_win.h"
#include <objbase.h>
#include <propkey.h>
#include <windows.devices.enumeration.h>
#include <windows.media.devices.h>
#include <algorithm>
#include <cmath>
#include <memory>
#include <utility>
#include "base/cxx17_backports.h"
#include "base/logging.h"
#include "base/metrics/histogram_functions.h"
#include "base/metrics/histogram_macros.h"
#include "base/strings/stringprintf.h"
#include "base/strings/utf_string_conversions.h"
#include "base/trace_event/trace_event.h"
#include "base/win/core_winrt_util.h"
#include "base/win/scoped_propvariant.h"
#include "base/win/scoped_variant.h"
#include "base/win/vector.h"
#include "base/win/windows_version.h"
#include "media/audio/audio_device_description.h"
#include "media/audio/audio_features.h"
#include "media/audio/win/avrt_wrapper_win.h"
#include "media/audio/win/core_audio_util_win.h"
#include "media/audio/win/volume_range_util.h"
#include "media/base/audio_block_fifo.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_timestamp_helper.h"
#include "media/base/channel_layout.h"
#include "media/base/limits.h"
#include "media/base/media_switches.h"
#include "media/base/timestamp_constants.h"
using ABI::Windows::Foundation::Collections::IVectorView;
using ABI::Windows::Media::Devices::IMediaDeviceStatics;
using ABI::Windows::Media::Effects::IAudioCaptureEffectsManager;
using ABI::Windows::Media::Effects::IAudioEffectsManagerStatics;
using base::win::GetActivationFactory;
using base::win::ScopedCoMem;
using base::win::ScopedCOMInitializer;
using base::win::ScopedHString;
using Microsoft::WRL::ComPtr;
namespace media {
namespace {
constexpr char kUwpDeviceIdPrefix[] = "\\\\?\\SWD#MMDEVAPI#";
constexpr uint32_t KSAUDIO_SPEAKER_UNSUPPORTED = 0;
// Converts a COM error into a human-readable string.
std::string ErrorToString(HRESULT hresult) {
return CoreAudioUtil::ErrorToString(hresult);
}
// Errors when initializing the audio client related to the audio format. Split
// by whether we're using format conversion or not. Used for reporting stats -
// do not renumber entries.
enum FormatRelatedInitError {
kUnsupportedFormat = 0,
kUnsupportedFormatWithFormatConversion = 1,
kInvalidArgument = 2,
kInvalidArgumentWithFormatConversion = 3,
kCount
};
bool IsSupportedFormatForConversion(WAVEFORMATEXTENSIBLE* format_ex) {
WAVEFORMATEX* format = &format_ex->Format;
if (format->nSamplesPerSec < limits::kMinSampleRate ||
format->nSamplesPerSec > limits::kMaxSampleRate) {
return false;
}
switch (format->wBitsPerSample) {
case 8:
case 16:
case 32:
break;
default:
return false;
}
if (GuessChannelLayout(format->nChannels) == CHANNEL_LAYOUT_UNSUPPORTED) {
LOG(ERROR) << "Hardware configuration not supported for audio conversion";
return false;
}
return true;
}
// Converts ChannelLayout to Microsoft's channel configuration but only discrete
// and up to stereo is supported currently. All other multi-channel layouts
// return KSAUDIO_SPEAKER_UNSUPPORTED.
ChannelConfig ChannelLayoutToChannelConfig(ChannelLayout layout) {
switch (layout) {
case CHANNEL_LAYOUT_DISCRETE:
return KSAUDIO_SPEAKER_DIRECTOUT;
case CHANNEL_LAYOUT_MONO:
return KSAUDIO_SPEAKER_MONO;
case CHANNEL_LAYOUT_STEREO:
return KSAUDIO_SPEAKER_STEREO;
default:
LOG(WARNING) << "Unsupported channel layout: " << layout;
// KSAUDIO_SPEAKER_UNSUPPORTED equals 0 and corresponds to "no specific
// channel order".
return KSAUDIO_SPEAKER_UNSUPPORTED;
}
}
const char* StreamOpenResultToString(
WASAPIAudioInputStream::StreamOpenResult result) {
switch (result) {
case WASAPIAudioInputStream::OPEN_RESULT_OK:
return "OK";
case WASAPIAudioInputStream::OPEN_RESULT_CREATE_INSTANCE:
return "CREATE_INSTANCE";
case WASAPIAudioInputStream::OPEN_RESULT_NO_ENDPOINT:
return "NO_ENDPOINT";
case WASAPIAudioInputStream::OPEN_RESULT_NO_STATE:
return "NO_STATE";
case WASAPIAudioInputStream::OPEN_RESULT_DEVICE_NOT_ACTIVE:
return "DEVICE_NOT_ACTIVE";
case WASAPIAudioInputStream::OPEN_RESULT_ACTIVATION_FAILED:
return "ACTIVATION_FAILED";
case WASAPIAudioInputStream::OPEN_RESULT_FORMAT_NOT_SUPPORTED:
return "FORMAT_NOT_SUPPORTED";
case WASAPIAudioInputStream::OPEN_RESULT_AUDIO_CLIENT_INIT_FAILED:
return "AUDIO_CLIENT_INIT_FAILED";
case WASAPIAudioInputStream::OPEN_RESULT_GET_BUFFER_SIZE_FAILED:
return "GET_BUFFER_SIZE_FAILED";
case WASAPIAudioInputStream::OPEN_RESULT_LOOPBACK_ACTIVATE_FAILED:
return "LOOPBACK_ACTIVATE_FAILED";
case WASAPIAudioInputStream::OPEN_RESULT_LOOPBACK_INIT_FAILED:
return "LOOPBACK_INIT_FAILED";
case WASAPIAudioInputStream::OPEN_RESULT_SET_EVENT_HANDLE:
return "SET_EVENT_HANDLE";
case WASAPIAudioInputStream::OPEN_RESULT_NO_CAPTURE_CLIENT:
return "NO_CAPTURE_CLIENT";
case WASAPIAudioInputStream::OPEN_RESULT_NO_AUDIO_VOLUME:
return "NO_AUDIO_VOLUME";
case WASAPIAudioInputStream::OPEN_RESULT_OK_WITH_RESAMPLING:
return "OK_WITH_RESAMPLING";
}
return "UNKNOWN";
}
const char* EffectTypeToString(
ABI::Windows::Media::Effects::AudioEffectType type) {
switch (type) {
case ABI::Windows::Media::Effects::AudioEffectType_Other:
return "Other/None";
case ABI::Windows::Media::Effects::AudioEffectType_AcousticEchoCancellation:
return "AcousticEchoCancellation";
case ABI::Windows::Media::Effects::AudioEffectType_NoiseSuppression:
return "NoiseSuppression";
case ABI::Windows::Media::Effects::AudioEffectType_AutomaticGainControl:
return "AutomaticGainControl";
case ABI::Windows::Media::Effects::AudioEffectType_BeamForming:
return "BeamForming";
case ABI::Windows::Media::Effects::AudioEffectType_ConstantToneRemoval:
return "ConstantToneRemoval";
case ABI::Windows::Media::Effects::AudioEffectType_Equalizer:
return "Equalizer";
case ABI::Windows::Media::Effects::AudioEffectType_LoudnessEqualizer:
return "LoudnessEqualizer";
case ABI::Windows::Media::Effects::AudioEffectType_BassBoost:
return "BassBoost";
case ABI::Windows::Media::Effects::AudioEffectType_VirtualSurround:
return "VirtualSurround";
case ABI::Windows::Media::Effects::AudioEffectType_VirtualHeadphones:
return "VirtualHeadphones";
case ABI::Windows::Media::Effects::AudioEffectType_SpeakerFill:
return "SpeakerFill";
case ABI::Windows::Media::Effects::AudioEffectType_RoomCorrection:
return "RoomCorrection";
case ABI::Windows::Media::Effects::AudioEffectType_BassManagement:
return "BassManagement";
case ABI::Windows::Media::Effects::AudioEffectType_EnvironmentalEffects:
return "EnvironmentalEffects";
case ABI::Windows::Media::Effects::AudioEffectType_SpeakerProtection:
return "SpeakerProtection";
case ABI::Windows::Media::Effects::AudioEffectType_SpeakerCompensation:
return "SpeakerCompensation";
case ABI::Windows::Media::Effects::AudioEffectType_DynamicRangeCompression:
return "DynamicRangeCompression";
}
return "Unknown";
}
bool VariantBoolToBool(VARIANT_BOOL var_bool) {
switch (var_bool) {
case VARIANT_TRUE:
return true;
case VARIANT_FALSE:
return false;
}
LOG(ERROR) << "Invalid VARIANT_BOOL type";
return false;
}
std::string GetOpenLogString(WASAPIAudioInputStream::StreamOpenResult result,
HRESULT hr,
WAVEFORMATEXTENSIBLE input_format,
WAVEFORMATEX output_format) {
return base::StringPrintf(
"WAIS::Open => (ERROR: result=%s, hresult=%#lx, input_format=[%s], "
"output_format=[%s])",
StreamOpenResultToString(result), hr,
CoreAudioUtil::WaveFormatToString(&input_format).c_str(),
CoreAudioUtil::WaveFormatToString(&output_format).c_str());
}
bool InitializeUWPSupport() {
// Place the actual body of the initialization in a lambda and store the
// result as a static since we don't expect this result to change between
// runs.
static const bool initialization_result = []() {
// Windows.Media.Effects and Windows.Media.Devices requires Windows 10 build
// 10.0.10240.0.
if (base::win::GetVersion() < base::win::Version::WIN10) {
DLOG(WARNING) << "AudioCaptureEffectsManager requires Windows 10";
return false;
}
DCHECK_GE(base::win::OSInfo::GetInstance()->version_number().build, 10240u);
// Provide access to Core WinRT/UWP functions and load all required HSTRING
// functions available from Win8 and onwards. ScopedHString is a wrapper
// around an HSTRING and it requires certain functions that need to be
// delayloaded to avoid breaking Chrome on Windows 7.
if (!(base::win::ResolveCoreWinRTDelayload() &&
base::win::ScopedHString::ResolveCoreWinRTStringDelayload())) {
// Failed loading functions from combase.dll.
DLOG(WARNING) << "Failed to initialize WinRT/UWP";
return false;
}
return true;
}();
return initialization_result;
}
} // namespace
WASAPIAudioInputStream::WASAPIAudioInputStream(
AudioManagerWin* manager,
const AudioParameters& params,
const std::string& device_id,
AudioManager::LogCallback log_callback)
: manager_(manager),
device_id_(device_id),
log_callback_(std::move(log_callback)) {
DCHECK(manager_);
DCHECK(!device_id_.empty());
DCHECK(!log_callback_.is_null());
DCHECK_LE(params.channels(), 2);
DCHECK(params.channel_layout() == CHANNEL_LAYOUT_MONO ||
params.channel_layout() == CHANNEL_LAYOUT_STEREO ||
params.channel_layout() == CHANNEL_LAYOUT_DISCRETE);
SendLogMessage("%s({device_id=%s}, {params=[%s]})", __func__,
device_id.c_str(), params.AsHumanReadableString().c_str());
// Load the Avrt DLL if not already loaded. Required to support MMCSS.
bool avrt_init = avrt::Initialize();
if (!avrt_init)
SendLogMessage("%s => (WARNING: failed to load Avrt.dll)", __func__);
const SampleFormat kSampleFormat = kSampleFormatS16;
// The clients asks for an input stream specified by |params|. Start by
// setting up an input device format according to the same specification.
// If all goes well during the upcoming initialization, this format will not
// change. However, under some circumstances, minor changes can be required
// to fit the current input audio device. If so, a FIFO and/or and audio
// converter might be needed to ensure that the output format of this stream
// matches what the client asks for.
WAVEFORMATEX* format = &input_format_.Format;
format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format->nChannels = params.channels();
format->nSamplesPerSec = params.sample_rate();
format->wBitsPerSample = SampleFormatToBitsPerChannel(kSampleFormat);
format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels;
format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign;
// Add the parts which are unique to WAVE_FORMAT_EXTENSIBLE which can be
// required in combination with e.g. multi-channel microphone arrays.
format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
input_format_.Samples.wValidBitsPerSample = format->wBitsPerSample;
input_format_.dwChannelMask =
ChannelLayoutToChannelConfig(params.channel_layout());
input_format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
SendLogMessage("%s => (audio engine format=[%s])", __func__,
CoreAudioUtil::WaveFormatToString(&input_format_).c_str());
// Set up the fixed output format based on |params|. Will not be changed and
// does not required an extended wave format structure since any multi-channel
// input will be converted to stereo.
output_format_.wFormatTag = WAVE_FORMAT_PCM;
output_format_.nChannels = format->nChannels;
output_format_.nSamplesPerSec = format->nSamplesPerSec;
output_format_.wBitsPerSample = format->wBitsPerSample;
output_format_.nBlockAlign = format->nBlockAlign;
output_format_.nAvgBytesPerSec = format->nAvgBytesPerSec;
output_format_.cbSize = 0;
SendLogMessage("%s => (audio sink format=[%s])", __func__,
CoreAudioUtil::WaveFormatToString(&output_format_).c_str());
// Size in bytes of each audio frame.
frame_size_bytes_ = format->nBlockAlign;
// Store size of audio packets which we expect to get from the audio
// endpoint device in each capture event.
packet_size_bytes_ = params.GetBytesPerBuffer(kSampleFormat);
packet_size_frames_ = packet_size_bytes_ / format->nBlockAlign;
SendLogMessage(
"%s => (packet size=[%zu bytes/%zu audio frames/%.3f milliseconds])",
__func__, packet_size_bytes_, packet_size_frames_,
params.GetBufferDuration().InMillisecondsF());
// 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_ready_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
DCHECK(audio_samples_ready_event_.IsValid());
// Create the event which will be set in Stop() when capturing shall stop.
stop_capture_event_.Set(CreateEvent(NULL, FALSE, FALSE, NULL));
DCHECK(stop_capture_event_.IsValid());
}
WASAPIAudioInputStream::~WASAPIAudioInputStream() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
}
AudioInputStream::OpenOutcome WASAPIAudioInputStream::Open() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
SendLogMessage("%s([opened=%s])", __func__, opened_ ? "true" : "false");
if (opened_) {
return OpenOutcome::kAlreadyOpen;
}
// Obtain a reference to the IMMDevice interface of the capturing device with
// the specified unique identifier or role which was set at construction.
HRESULT hr = SetCaptureDevice();
if (FAILED(hr)) {
ReportOpenResult(hr);
return OpenOutcome::kFailed;
}
// Check if raw audio processing is supported for the selected capture device.
raw_processing_supported_ = RawProcessingSupported();
if (raw_processing_supported_ &&
!AudioDeviceDescription::IsLoopbackDevice(device_id_) &&
InitializeUWPSupport()) {
// Retrieve a unique identifier of the selected audio device but in a
// format which can be used by UWP (or Core WinRT) APIs. It can then be
// utilized in combination with the Windows.Media.Effects UWP API to
// discover the audio processing chain on a device.
std::string uwp_device_id = GetUWPDeviceId();
if (!uwp_device_id.empty()) {
// For the selected device, generate two lists of enabled audio effects
// and store them in |default_effect_types_| and |raw_effect_types_|.
// Default corresponds to "Normal audio signal processing" and Raw is for
// "Minimal audio signal processing". These two lists are used for UMA
// stats when the stream is closed.
GetAudioCaptureEffects(uwp_device_id);
}
}
// Obtain an IAudioClient interface which enables us to create and initialize
// an audio stream between an audio application and the audio engine.
hr = endpoint_device_->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr,
&audio_client_);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_ACTIVATION_FAILED;
ReportOpenResult(hr);
return OpenOutcome::kFailed;
}
#ifndef NDEBUG
// Retrieve the stream format which the audio engine uses for its internal
// processing/mixing of shared-mode streams. This function call is for
// diagnostic purposes only and only in debug mode.
hr = GetAudioEngineStreamFormat();
#endif
// Attempt to enable communications category and raw capture mode on the audio
// stream. Ignoring return value since the method logs its own error messages
// and it should be OK to continue opening the stream even after a failure.
if (base::FeatureList::IsEnabled(media::kWasapiRawAudioCapture) &&
raw_processing_supported_ &&
!AudioDeviceDescription::IsLoopbackDevice(device_id_)) {
SetCommunicationsCategoryAndRawCaptureMode();
}
// Verify that the selected audio endpoint supports the specified format
// set during construction and using the specified client properties.
hr = S_OK;
if (!DesiredFormatIsSupported(&hr)) {
open_result_ = OPEN_RESULT_FORMAT_NOT_SUPPORTED;
ReportOpenResult(hr);
return OpenOutcome::kFailed;
}
// Initialize the audio stream between the client and the device using
// shared mode and a lowest possible glitch-free latency.
hr = InitializeAudioEngine();
if (SUCCEEDED(hr) && converter_)
open_result_ = OPEN_RESULT_OK_WITH_RESAMPLING;
ReportOpenResult(hr); // Report before we assign a value to |opened_|.
opened_ = SUCCEEDED(hr);
if (opened_) {
return OpenOutcome::kSuccess;
}
switch (hr) {
case E_ACCESSDENIED:
return OpenOutcome::kFailedSystemPermissions;
case AUDCLNT_E_DEVICE_IN_USE:
return OpenOutcome::kFailedInUse;
default:
return OpenOutcome::kFailed;
}
}
void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
DCHECK(callback);
SendLogMessage("%s([opened=%s, started=%s])", __func__,
opened_ ? "true" : "false", started_ ? "true" : "false");
if (!opened_)
return;
if (started_)
return;
// Check if the master volume level of the opened audio session is set to
// zero and store the information for a UMA histogram generated in Stop().
// Valid volume levels are in the range 0.0 to 1.0.
// See http://crbug.com/1014443 for details why this is needed.
if (GetVolume() == 0.0) {
SendLogMessage("%s => (WARNING: Input audio session starts at zero volume)",
__func__);
audio_session_starts_at_zero_volume_ = true;
}
if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId &&
system_audio_volume_) {
BOOL muted = false;
system_audio_volume_->GetMute(&muted);
// If the system audio is muted at the time of capturing, then no need to
// mute it again, and later we do not unmute system audio when stopping
// capturing.
if (!muted) {
system_audio_volume_->SetMute(true, nullptr);
mute_done_ = true;
}
}
DCHECK(!sink_);
sink_ = callback;
// Starts periodic AGC microphone measurements if the AGC has been enabled
// using SetAutomaticGainControl().
StartAgc();
// Create and start the thread that will drive the capturing by waiting for
// capture events.
DCHECK(!capture_thread_.get());
capture_thread_ = std::make_unique<base::DelegateSimpleThread>(
this, "wasapi_capture_thread",
base::SimpleThread::Options(base::ThreadPriority::REALTIME_AUDIO));
capture_thread_->Start();
// Start streaming data between the endpoint buffer and the audio engine.
HRESULT hr = audio_client_->Start();
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioClient::Start=[%s])", __func__,
ErrorToString(hr).c_str());
}
if (SUCCEEDED(hr) && audio_render_client_for_loopback_.Get()) {
hr = audio_render_client_for_loopback_->Start();
if (FAILED(hr))
SendLogMessage("%s => (ERROR: IAudioClient::Start=[%s] (loopback))",
__func__, ErrorToString(hr).c_str());
}
started_ = SUCCEEDED(hr);
}
void WASAPIAudioInputStream::Stop() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
SendLogMessage("%s([started=%s])", __func__, started_ ? "true" : "false");
if (!started_)
return;
// Only upload UMA histogram for the case when AGC is enabled, i.e., for
// WebRTC based audio input streams.
const bool add_uma_histogram = GetAutomaticGainControl();
// We have muted system audio for capturing, so we need to unmute it when
// capturing stops.
if (device_id_ == AudioDeviceDescription::kLoopbackWithMuteDeviceId &&
mute_done_) {
DCHECK(system_audio_volume_);
if (system_audio_volume_) {
system_audio_volume_->SetMute(false, nullptr);
mute_done_ = false;
}
}
absl::optional<VolumeRange> volume_range;
if (add_uma_histogram && system_audio_volume_ &&
!AudioDeviceDescription::IsLoopbackDevice(device_id_)) {
VolumeRange range;
HRESULT hr = system_audio_volume_->GetVolumeRange(
&range.min_volume_db, &range.max_volume_db, &range.volume_step_db);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioEndpointVolume::GetVolumeRange=[%s])",
__func__, ErrorToString(hr).c_str());
} else {
SendLogMessage("%s => (IAudioEndpointVolume::GetVolumeRange) %f %f %f",
__func__, range.min_volume_db, range.max_volume_db,
range.volume_step_db);
volume_range = range;
}
}
// Stops periodic AGC microphone measurements.
StopAgc();
// Shut down the capture thread.
if (stop_capture_event_.IsValid()) {
SetEvent(stop_capture_event_.Get());
}
// Stop the input audio streaming.
HRESULT hr = audio_client_->Stop();
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioClient::Stop=[%s])", __func__,
ErrorToString(hr).c_str());
}
// Wait until the thread completes and perform cleanup.
if (capture_thread_) {
SetEvent(stop_capture_event_.Get());
capture_thread_->Join();
capture_thread_.reset();
}
// Upload UMA histogram to track down possible issue that can lead to a
// "no audio" state. See http://crbug.com/1014443.
if (add_uma_histogram) {
base::UmaHistogramBoolean("Media.Audio.InputVolumeStartsAtZeroWin",
audio_session_starts_at_zero_volume_);
audio_session_starts_at_zero_volume_ = false;
LogVolumeRangeUmaHistograms(volume_range);
}
SendLogMessage(
"%s => (timestamp(n)-timestamp(n-1)=[min: %.3f msec, max: %.3f msec])",
__func__, min_timestamp_diff_.InMillisecondsF(),
max_timestamp_diff_.InMillisecondsF());
started_ = false;
sink_ = nullptr;
}
void WASAPIAudioInputStream::Close() {
SendLogMessage("%s()", __func__);
// It is valid to call Close() before calling open or Start().
// It is also valid to call Close() after Start() has been called.
Stop();
// Only upload UMA histogram for the case when AGC is enabled, i.e., for
// WebRTC based audio input streams.
if (GetAutomaticGainControl()) {
// Upload UMA histogram to track if the capture device supported raw audio
// capture or not. See https://crbug.com/1133643.
base::UmaHistogramBoolean("Media.Audio.RawProcessingSupportedWin",
raw_processing_supported_);
for (auto const& type : default_effect_types_) {
base::UmaHistogramSparse("Media.Audio.Capture.Win.DefaultEffectType",
type);
SendLogMessage("%s => (Media.Audio.Capture.Win.DefaultEffectType=%s)",
__func__, EffectTypeToString(type));
}
for (auto const& type : raw_effect_types_) {
base::UmaHistogramSparse("Media.Audio.Capture.Win.RawEffectType", type);
SendLogMessage("%s => (Media.Audio.Capture.Win.RawEffectType=%s)",
__func__, EffectTypeToString(type));
}
}
if (converter_)
converter_->RemoveInput(this);
ReportAndResetGlitchStats();
// Inform the audio manager that we have been closed. This will cause our
// destruction.
manager_->ReleaseInputStream(this);
}
double WASAPIAudioInputStream::GetMaxVolume() {
// Verify that Open() has been called successfully, to ensure that an audio
// session exists and that an ISimpleAudioVolume interface has been created.
DLOG_IF(ERROR, !opened_) << "Open() has not been called successfully";
if (!opened_)
return 0.0;
// The effective volume value is always in the range 0.0 to 1.0, hence
// we can return a fixed value (=1.0) here.
return 1.0;
}
void WASAPIAudioInputStream::SetVolume(double volume) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
DCHECK_GE(volume, 0.0);
DCHECK_LE(volume, 1.0);
SendLogMessage("%s({volume=%.2f} [opened=%s])", __func__, volume,
opened_ ? "true" : "false");
if (!opened_)
return;
// Set a new master volume level. Valid volume levels are in the range
// 0.0 to 1.0. Ignore volume-change events.
HRESULT hr = simple_audio_volume_->SetMasterVolume(static_cast<float>(volume),
nullptr);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: ISimpleAudioVolume::SetMasterVolume=[%s])",
__func__, ErrorToString(hr).c_str());
}
// Update the AGC volume level based on the last setting above. Note that,
// the volume-level resolution is not infinite and it is therefore not
// possible to assume that the volume provided as input parameter can be
// used directly. Instead, a new query to the audio hardware is required.
// This method does nothing if AGC is disabled.
UpdateAgcVolume();
}
double WASAPIAudioInputStream::GetVolume() {
DCHECK(opened_) << "Open() has not been called successfully";
if (!opened_)
return 0.0;
// Retrieve the current volume level. The value is in the range 0.0 to 1.0.
float level = 0.0f;
HRESULT hr = simple_audio_volume_->GetMasterVolume(&level);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: ISimpleAudioVolume::GetMasterVolume=[%s])",
__func__, ErrorToString(hr).c_str());
}
return static_cast<double>(level);
}
bool WASAPIAudioInputStream::IsMuted() {
DCHECK(opened_) << "Open() has not been called successfully";
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
if (!opened_)
return false;
// Retrieves the current muting state for the audio session.
BOOL is_muted = FALSE;
HRESULT hr = simple_audio_volume_->GetMute(&is_muted);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: ISimpleAudioVolume::GetMute=[%s])", __func__,
ErrorToString(hr).c_str());
}
return is_muted != FALSE;
}
void WASAPIAudioInputStream::SetOutputDeviceForAec(
const std::string& output_device_id) {
// Not supported. Do nothing.
}
void WASAPIAudioInputStream::SendLogMessage(const char* format, ...) {
if (log_callback_.is_null())
return;
va_list args;
va_start(args, format);
std::string msg("WAIS::" + base::StringPrintV(format, args));
log_callback_.Run(msg);
DVLOG(1) << msg;
va_end(args);
}
void WASAPIAudioInputStream::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(ERROR) << "WAIS::" << __func__
<< " => (ERROR: Failed to enable MMCSS (error code=" << err
<< "))";
}
// Allocate a buffer with a size that enables us to take care of cases like:
// 1) The recorded buffer size is smaller, or does not match exactly with,
// the selected packet size used in each callback.
// 2) The selected buffer size is larger than the recorded buffer size in
// each event.
// In the case where no resampling is required, a single buffer should be
// enough but in case we get buffers that don't match exactly, we'll go with
// two. Same applies if we need to resample and the buffer ratio is perfect.
// However if the buffer ratio is imperfect, we will need 3 buffers to safely
// be able to buffer up data in cases where a conversion requires two audio
// buffers (and we need to be able to write to the third one).
size_t capture_buffer_size =
std::max(2 * endpoint_buffer_size_frames_ * frame_size_bytes_,
2 * packet_size_frames_ * frame_size_bytes_);
int buffers_required = capture_buffer_size / packet_size_bytes_;
if (converter_ && imperfect_buffer_size_conversion_)
++buffers_required;
DCHECK(!fifo_);
fifo_ = std::make_unique<AudioBlockFifo>(
input_format_.Format.nChannels, packet_size_frames_, buffers_required);
DVLOG(1) << "AudioBlockFifo buffer count: " << buffers_required;
bool recording = true;
bool error = false;
HANDLE wait_array[2] = {stop_capture_event_.Get(),
audio_samples_ready_event_.Get()};
record_start_time_ = base::TimeTicks::Now();
last_capture_time_ = base::TimeTicks();
max_timestamp_diff_ = base::TimeDelta::Min();
min_timestamp_diff_ = base::TimeDelta::Max();
while (recording && !error) {
// Wait for a close-down event or a new capture event.
DWORD wait_result = WaitForMultipleObjects(2, wait_array, FALSE, INFINITE);
switch (wait_result) {
case WAIT_OBJECT_0 + 0:
// |stop_capture_event_| has been set.
recording = false;
break;
case WAIT_OBJECT_0 + 1:
// |audio_samples_ready_event_| has been set.
PullCaptureDataAndPushToSink();
break;
case WAIT_FAILED:
default:
error = true;
break;
}
}
if (recording && error) {
// TODO(henrika): perhaps it worth improving the cleanup here by e.g.
// stopping the audio client, joining the thread etc.?
NOTREACHED() << "WASAPI capturing failed with error code "
<< GetLastError();
}
// Disable MMCSS.
if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
PLOG(WARNING) << "Failed to disable MMCSS";
}
fifo_.reset();
}
void WASAPIAudioInputStream::PullCaptureDataAndPushToSink() {
TRACE_EVENT1("audio", "WASAPIAudioInputStream::PullCaptureDataAndPushToSink",
"sample rate", input_format_.Format.nSamplesPerSec);
UINT64 last_device_position = 0;
UINT32 num_frames_in_next_packet = 0;
// Get the number of frames in the next data packet in the capture endpoint
// buffer. The count reported by GetNextPacketSize matches the count retrieved
// in the GetBuffer call that follows this call.
HRESULT hr =
audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
if (FAILED(hr)) {
LOG(ERROR) << "WAIS::" << __func__
<< " => (ERROR: 1-IAudioCaptureClient::GetNextPacketSize=["
<< ErrorToString(hr).c_str() << "])";
return;
}
// Pull data from the capture endpoint buffer until it's empty or an error
// occurs. Drains the WASAPI capture buffer fully.
while (num_frames_in_next_packet > 0) {
BYTE* data_ptr = nullptr;
UINT32 num_frames_to_read = 0;
DWORD flags = 0;
UINT64 device_position = 0;
UINT64 capture_time_100ns = 0;
// Retrieve the amount of data in the capture endpoint buffer, replace it
// with silence if required, create callbacks for each packet and store
// non-delivered data for the next event.
hr =
audio_capture_client_->GetBuffer(&data_ptr, &num_frames_to_read, &flags,
&device_position, &capture_time_100ns);
if (hr == AUDCLNT_S_BUFFER_EMPTY) {
DCHECK_EQ(num_frames_to_read, 0u);
return;
}
if (hr == AUDCLNT_E_OUT_OF_ORDER) {
// A previous IAudioCaptureClient::GetBuffer() call is still in effect.
// Release any acquired buffer to be able to try reading a buffer again.
audio_capture_client_->ReleaseBuffer(num_frames_to_read);
}
if (FAILED(hr)) {
LOG(ERROR) << "WAIS::" << __func__
<< " => (ERROR: IAudioCaptureClient::GetBuffer=["
<< ErrorToString(hr).c_str() << "])";
return;
}
// The data in the packet is not correlated with the previous packet's
// device position; this is possibly due to a stream state transition or
// timing glitch. Note that, usage of this flag was added after the existing
// glitch detection in UpdateGlitchCount() and it will be used as a
// supplementary scheme initially.
// The behavior of the AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY flag is
// undefined on the application's first call to GetBuffer after Start and
// Windows 7 or later is required for support.
if (device_position > 0 && flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) {
LOG(WARNING) << "WAIS::" << __func__
<< " => (WARNING: AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)";
++num_data_discontinuity_warnings_;
}
// The time at which the device's stream position was recorded is uncertain.
// Thus, the client might be unable to accurately set a time stamp for the
// current data packet.
bool timestamp_error_was_detected = false;
if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR) {
// TODO(https://crbug.com/825744): it might be possible to improve error
// handling here and avoid using the counter in |capture_time_100ns|.
LOG(WARNING) << "WAIS::" << __func__
<< " => (WARNING: AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR)";
if (num_timestamp_errors_ == 0) {
// Measure the time it took until the first timestamp error was found.
time_until_first_timestamp_error_ =
base::TimeTicks::Now() - record_start_time_;
}
++num_timestamp_errors_;
timestamp_error_was_detected = true;
}
// If the device position has changed, we assume this data belongs to a new
// chunk, so we report delay and glitch stats and update the last and next
// expected device positions.
// If the device position has not changed we assume this data belongs to the
// previous chunk, and only update the expected next device position.
if (device_position != last_device_position) {
UpdateGlitchCount(device_position);
last_device_position = device_position;
expected_next_device_position_ = device_position + num_frames_to_read;
} else {
expected_next_device_position_ += num_frames_to_read;
}
base::TimeTicks capture_time;
if (!timestamp_error_was_detected) {
// Use the latest |capture_time_100ns| since it is marked as valid.
capture_time += base::Microseconds(capture_time_100ns / 10.0);
}
if (capture_time <= last_capture_time_) {
// Latest |capture_time_100ns| can't be trusted. Ensure a monotonic time-
// stamp sequence by adding one microsecond to the latest timestamp.
capture_time = last_capture_time_ + base::Microseconds(1);
}
// Keep track of max and min time difference between two successive time-
// stamps. Results are used in Stop() to verify that the time-stamp sequence
// was monotonic.
if (!last_capture_time_.is_null()) {
const auto delta_ts = capture_time - last_capture_time_;
DCHECK_GT(device_position, 0u);
DCHECK_GT(delta_ts, base::TimeDelta::Min());
if (delta_ts > max_timestamp_diff_) {
max_timestamp_diff_ = delta_ts;
} else if (delta_ts < min_timestamp_diff_) {
min_timestamp_diff_ = delta_ts;
}
}
// Store the capture timestamp. Might be used as reference next time if
// a new valid timestamp can't be retrieved to always guarantee a monotonic
// sequence.
last_capture_time_ = capture_time;
// Adjust |capture_time| for the FIFO before pushing.
capture_time -= AudioTimestampHelper::FramesToTime(
fifo_->GetAvailableFrames(), input_format_.Format.nSamplesPerSec);
if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
fifo_->PushSilence(num_frames_to_read);
} else {
fifo_->Push(data_ptr, num_frames_to_read,
input_format_.Format.wBitsPerSample / 8);
}
hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
if (FAILED(hr)) {
LOG(ERROR) << "WAIS::" << __func__
<< " => (ERROR: IAudioCaptureClient::ReleaseBuffer=["
<< ErrorToString(hr).c_str() << "])";
return;
}
// Get a cached AGC volume level which is updated once every second on the
// audio manager thread. Note that, |volume| is also updated each time
// SetVolume() is called through IPC by the render-side AGC.
double volume = 0.0;
GetAgcVolume(&volume);
// Deliver captured data to the registered consumer using a packet size
// which was specified at construction.
while (fifo_->available_blocks()) {
if (converter_) {
if (imperfect_buffer_size_conversion_ &&
fifo_->available_blocks() == 1) {
// Special case. We need to buffer up more audio before we can convert
// or else we'll suffer an underrun.
// TODO(grunell): Verify this is really true.
return;
}
converter_->Convert(convert_bus_.get());
sink_->OnData(convert_bus_.get(), capture_time, volume);
// Move the capture time forward for each vended block.
capture_time += AudioTimestampHelper::FramesToTime(
convert_bus_->frames(), output_format_.nSamplesPerSec);
} else {
sink_->OnData(fifo_->Consume(), capture_time, volume);
// Move the capture time forward for each vended block.
capture_time += AudioTimestampHelper::FramesToTime(
packet_size_frames_, input_format_.Format.nSamplesPerSec);
}
}
// Get the number of frames in the next data packet in the capture endpoint
// buffer. Keep reading if more samples exist.
hr = audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
if (FAILED(hr)) {
LOG(ERROR) << "WAIS::" << __func__
<< " => (ERROR: 2-IAudioCaptureClient::GetNextPacketSize=["
<< ErrorToString(hr).c_str() << "])";
return;
}
} // while (num_frames_in_next_packet > 0)
}
void WASAPIAudioInputStream::HandleError(HRESULT err) {
NOTREACHED() << "Error code: " << err;
if (sink_)
sink_->OnError();
}
HRESULT WASAPIAudioInputStream::SetCaptureDevice() {
DCHECK_EQ(OPEN_RESULT_OK, open_result_);
DCHECK(!endpoint_device_.Get());
SendLogMessage("%s()", __func__);
Microsoft::WRL::ComPtr<IMMDeviceEnumerator> enumerator;
HRESULT hr = ::CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr,
CLSCTX_ALL, IID_PPV_ARGS(&enumerator));
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_CREATE_INSTANCE;
return hr;
}
// Retrieve the IMMDevice by using the specified role or the specified
// unique endpoint device-identification string.
// To open a stream in loopback mode, the client must obtain an IMMDevice
// interface for the rendering endpoint device. Make that happen if needed;
// otherwise use default capture data-flow direction.
const EDataFlow data_flow =
AudioDeviceDescription::IsLoopbackDevice(device_id_) ? eRender : eCapture;
// Determine selected role to be used if the device is a default device.
const ERole role = AudioDeviceDescription::IsCommunicationsDevice(device_id_)
? eCommunications
: eConsole;
if (AudioDeviceDescription::IsDefaultDevice(device_id_) ||
AudioDeviceDescription::IsCommunicationsDevice(device_id_) ||
AudioDeviceDescription::IsLoopbackDevice(device_id_)) {
hr =
enumerator->GetDefaultAudioEndpoint(data_flow, role, &endpoint_device_);
} else {
hr = enumerator->GetDevice(base::UTF8ToWide(device_id_).c_str(),
&endpoint_device_);
}
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_NO_ENDPOINT;
return hr;
}
// Get the volume interface for the endpoint. Used in `Stop()` to query the
// volume range of the selected input device or to get/set mute state in
// `Start()` and `Stop()` if a loopback device with muted system audio is
// requested.
hr = endpoint_device_->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL,
nullptr, &system_audio_volume_);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_ACTIVATION_FAILED;
return hr;
}
// Verify that the audio endpoint device is active, i.e., the audio
// adapter that connects to the endpoint device is present and enabled.
DWORD state = DEVICE_STATE_DISABLED;
hr = endpoint_device_->GetState(&state);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_NO_STATE;
return hr;
}
if (!(state & DEVICE_STATE_ACTIVE)) {
DLOG(ERROR) << "Selected capture device is not active.";
open_result_ = OPEN_RESULT_DEVICE_NOT_ACTIVE;
hr = E_ACCESSDENIED;
}
return hr;
}
HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() {
HRESULT hr = S_OK;
#ifndef NDEBUG
base::win::ScopedCoMem<WAVEFORMATEX> format;
hr = audio_client_->GetMixFormat(&format);
if (FAILED(hr))
return hr;
DVLOG(1) << CoreAudioUtil::WaveFormatToString(format.get());
#endif
return hr;
}
bool WASAPIAudioInputStream::RawProcessingSupported() {
DCHECK(endpoint_device_.Get());
// Check if System.Devices.AudioDevice.RawProcessingSupported can be found
// and queried in the Windows Property System. It corresponds to raw
// processing mode support for the specified audio device. If its value is
// VARIANT_TRUE the device supports raw processing mode.
bool raw_processing_supported = false;
Microsoft::WRL::ComPtr<IPropertyStore> properties;
base::win::ScopedPropVariant raw_processing;
if (FAILED(endpoint_device_->OpenPropertyStore(STGM_READ, &properties)) ||
FAILED(
properties->GetValue(PKEY_Devices_AudioDevice_RawProcessingSupported,
raw_processing.Receive())) ||
raw_processing.get().vt != VT_BOOL) {
SendLogMessage(
"%s => (WARNING: failed to access "
"System.Devices.AudioDevice.RawProcessingSupported)",
__func__);
} else {
raw_processing_supported = VariantBoolToBool(raw_processing.get().boolVal);
SendLogMessage(
"%s => (System.Devices.AudioDevice.RawProcessingSupported=%s)",
__func__, raw_processing_supported ? "true" : "false");
}
return raw_processing_supported;
}
std::string WASAPIAudioInputStream::GetUWPDeviceId() {
DCHECK(endpoint_device_.Get());
// The Windows.Media.Devices.IMediaDeviceStatics interface provides access to
// the implementation of Windows.Media.Devices.MediaDevice.
ComPtr<IMediaDeviceStatics> media_device_statics;
HRESULT hr =
GetActivationFactory<IMediaDeviceStatics,
RuntimeClass_Windows_Media_Devices_MediaDevice>(
&media_device_statics);
if (FAILED(hr)) {
DLOG(ERROR) << "IMediaDeviceStatics factory failed: " << ErrorToString(hr);
return std::string();
}
// The remaining part of this method builds up the unique device ID needed
// by the Windows.Media.Effects.AudioEffectsManager UWP API to enumerate
// active capture effects like AEC and NS. The ID contains three parts.
// Example:
// 1) \\?\SWD#MMDEVAPI#
// 2) {0.0.1.00000000}.{7c24467c-94fc-4fa1-a2b2-a3f5d9cb8a5b}
// 3) #{2eef81be-33fa-4800-9670-1cd474972c3f}
// Where (1) is a constant string, (2) comes from the IMMDevice::GetId() API,
// and (3) is a substring of of the selector string which can be retrieved by
// the IMediaDeviceStatics::GetAudioCaptureSelector UWP API. Knowledge about
// the structure of this device ID can be gained by using the
// IMediaDeviceStatics::GetDefaultAudioCaptureId UWP API but this method also
// adds support for non default devices.
// (1) Start building the final device ID. Start with the constant prefix.
std::string device_id(kUwpDeviceIdPrefix);
// (2) Next, add the unique ID from IMMDevice::GetId() API.
// Example: {0.0.1.00000000}.{7c24467c-94fc-4fa1-a2b2-a3f5d9cb8a5b}.
ScopedCoMem<WCHAR> immdevice_id16;
hr = endpoint_device_->GetId(&immdevice_id16);
if (FAILED(hr)) {
DLOG(ERROR) << "IMMDevice::GetId failed: " << ErrorToString(hr);
return std::string();
}
std::string immdevice_id8;
base::WideToUTF8(immdevice_id16, wcslen(immdevice_id16), &immdevice_id8);
device_id.append(immdevice_id8);
// (3) Finally, add the last part from the selector string.
// Example: '#{2eef81be-33fa-4800-9670-1cd474972c3f}'.
HSTRING selector;
// Returns the identifier string of a device for capturing audio. A substring
// will be used when generating the final unique device ID.
// Example: part of the selector string can look like
// System.Devices.InterfaceClassGuid:="{2eef81be-33fa-4800-9670-1cd474972c3f}"
// and we want the {2eef81be-33fa-4800-9670-1cd474972c3f} substring for our
// purposes.
hr = media_device_statics->GetAudioCaptureSelector(&selector);
if (FAILED(hr)) {
DLOG(ERROR) << "IMediaDeviceStatics::GetAudioCaptureSelector failed: "
<< ErrorToString(hr);
return std::string();
}
device_id.append("#");
std::string selector_string = ScopedHString(selector).GetAsUTF8();
std::size_t start = selector_string.find("{");
std::size_t stop = selector_string.find("}", start + 1);
if (start != std::string::npos && stop != std::string::npos) {
// Will extract '{2eef81be-33fa-4800-9670-1cd474972c3f}' in the example
// above.
device_id.append(selector_string.substr(start, stop - start + 1));
} else {
DLOG(ERROR) << "Failed to extract System.Devices.InterfaceClassGuid string";
return std::string();
}
return device_id;
}
HRESULT WASAPIAudioInputStream::GetAudioCaptureEffects(
const std::string& uwp_device_id) {
DCHECK(!AudioDeviceDescription::IsLoopbackDevice(device_id_));
DCHECK(raw_processing_supported_);
DCHECK(!uwp_device_id.empty());
SendLogMessage("%s()", __func__);
// The Windows.Media.Effects.IAudioEffectsManagerStatics interface provides
// access to the implementation of Windows.Media.Effects.AudioEffectsManager.
ComPtr<IAudioEffectsManagerStatics> audio_effects_manager;
HRESULT hr = GetActivationFactory<
IAudioEffectsManagerStatics,
RuntimeClass_Windows_Media_Effects_AudioEffectsManager>(
&audio_effects_manager);
if (FAILED(hr)) {
SendLogMessage(
"%s => (ERROR: IAudioEffectsManagerStatics factory failed: [%s])",
__func__, ErrorToString(hr).c_str());
return hr;
}
SendLogMessage("%s => (uwp_device_id=[%s])", __func__, uwp_device_id.c_str());
ScopedHString device_id = ScopedHString::Create(uwp_device_id);
// Check capture effects for two different audio processing modes:
// - Default: Normal audio signal processing
// - Raw: Minimal audio signal processing
// Raw is included since it is not possible to disable all effects on all
// devices. In most cases, the number of found capture effects will be zero
// for the raw mode.
ABI::Windows::Media::AudioProcessing audio_processing_mode[] = {
ABI::Windows::Media::AudioProcessing::AudioProcessing_Default,
ABI::Windows::Media::AudioProcessing::AudioProcessing_Raw};
for (size_t i = 0; i < base::size(audio_processing_mode); ++i) {
// Create an AudioCaptureEffectsManager manager which can be used to
// discover the audio processing chain on a device for a specific media
// category and audio processing mode. The media category is fixed and set
// to Communications since that is what we aim at using when audio effects
// later are disabled.
ComPtr<IAudioCaptureEffectsManager> capture_effects_manager;
hr = audio_effects_manager->CreateAudioCaptureEffectsManagerWithMode(
device_id.get(),
ABI::Windows::Media::Capture::MediaCategory::
MediaCategory_Communications,
audio_processing_mode[i], &capture_effects_manager);
if (FAILED(hr)) {
SendLogMessage(
"%s => (ERROR: IAudioEffectsManagerStatics::"
"CreateAudioCaptureEffectsManager=[%s])",
__func__, ErrorToString(hr).c_str());
return hr;
}
// Get a list of audio effects on the device. Based on tests on different
// devices, only enabled effects will be included. Hence, if a user has
// explicitly disabled an effect using the System Sound Settings, that
// component will not show up here.
ComPtr<IVectorView<ABI::Windows::Media::Effects::AudioEffect*>> effects;
hr = capture_effects_manager->GetAudioCaptureEffects(&effects);
if (FAILED(hr)) {
SendLogMessage(
"%s => (ERROR: IAudioCaptureEffectsManager::"
"GetAudioCaptureEffects=[%s])",
__func__, ErrorToString(hr).c_str());
return hr;
}
unsigned int count = 0;
if (effects) {
// Returns number of supported effects.
effects->get_Size(&count);
}
// Store all supported and active effect types in |default_effect_types_|
// or |raw_effect_types_| depending on selected audio processing mode.
// These will be utilized later for UMA histograms.
for (unsigned int j = 0; j < count; ++j) {
ComPtr<ABI::Windows::Media::Effects::IAudioEffect> effect;
hr = effects->GetAt(j, &effect);
if (SUCCEEDED(hr)) {
ABI::Windows::Media::Effects::AudioEffectType type;
hr = effect->get_AudioEffectType(&type);
if (SUCCEEDED(hr)) {
audio_processing_mode[i] ==
ABI::Windows::Media::AudioProcessing::AudioProcessing_Default
? default_effect_types_.push_back(type)
: raw_effect_types_.push_back(type);
}
}
}
// For cases when no audio effects were found (common in raw mode), add a
// dummy effect type called AudioEffectType_Other so that the vector
// contains at least one value. This is done to ensure that an UMA histogram
// is uploaded also for the empty case. Hence, AudioEffectType_Other is
// used to indicate an unknown audio effect and "no audio effect found".
if (count == 0) {
const ABI::Windows::Media::Effects::AudioEffectType no_effect_found =
ABI::Windows::Media::Effects::AudioEffectType::AudioEffectType_Other;
audio_processing_mode[i] ==
ABI::Windows::Media::AudioProcessing::AudioProcessing_Default
? default_effect_types_.push_back(no_effect_found)
: raw_effect_types_.push_back(no_effect_found);
}
}
return hr;
}
HRESULT WASAPIAudioInputStream::SetCommunicationsCategoryAndRawCaptureMode() {
DCHECK(audio_client_.Get());
DCHECK(!AudioDeviceDescription::IsLoopbackDevice(device_id_));
DCHECK(raw_processing_supported_);
SendLogMessage("%s()", __func__);
Microsoft::WRL::ComPtr<IAudioClient2> audio_client2;
HRESULT hr = audio_client_.As(&audio_client2);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioClient2 is not supported)", __func__);
return hr;
}
// Use IAudioClient2::SetClientProperties() to set communications category
// and to enable raw stream capture if it is supported.
if (audio_client2.Get()) {
AudioClientProperties audio_props = {0};
audio_props.cbSize = sizeof(AudioClientProperties);
audio_props.bIsOffload = false;
// AudioCategory_Communications opts us in to communications policy and
// communications processing. AUDCLNT_STREAMOPTIONS_RAW turns off the
// processing, but not the policy.
audio_props.eCategory = AudioCategory_Communications;
// The audio stream is a 'raw' stream that bypasses all signal processing
// except for endpoint specific, always-on processing in the Audio
// Processing Object (APO), driver, and hardware.
audio_props.Options = AUDCLNT_STREAMOPTIONS_RAW;
hr = audio_client2->SetClientProperties(&audio_props);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioClient2::SetClientProperties=[%s])",
__func__, ErrorToString(hr).c_str());
}
}
return hr;
}
bool WASAPIAudioInputStream::DesiredFormatIsSupported(HRESULT* hr) {
SendLogMessage("%s()", __func__);
// An application that uses WASAPI to manage shared-mode streams can rely
// on the audio engine to perform only limited format conversions. The audio
// engine can convert between a standard PCM sample size used by the
// application and the floating-point samples that the engine uses for its
// internal processing. However, the format for an application stream
// typically must have the same number of channels and the same sample
// rate as the stream format used by the device.
// Many audio devices support both PCM and non-PCM stream formats. However,
// the audio engine can mix only PCM streams.
base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
HRESULT hresult = audio_client_->IsFormatSupported(
AUDCLNT_SHAREMODE_SHARED,
reinterpret_cast<const WAVEFORMATEX*>(&input_format_), &closest_match);
if (FAILED(hresult)) {
SendLogMessage("%s => (ERROR: IAudioClient::IsFormatSupported=[%s])",
__func__, ErrorToString(hresult).c_str());
}
if (hresult == S_FALSE) {
SendLogMessage(
"%s => (WARNING: Format is not supported but a closest match exists)",
__func__);
// Change the format we're going to ask for to better match with what the OS
// can provide. If we succeed in initializing the audio client in this
// format and are able to convert from this format, we will do that
// conversion.
WAVEFORMATEX* input_format = &input_format_.Format;
input_format->nChannels = closest_match->nChannels;
input_format->nSamplesPerSec = closest_match->nSamplesPerSec;
// If the closest match is fixed point PCM (WAVE_FORMAT_PCM or
// KSDATAFORMAT_SUBTYPE_PCM), we use the closest match's bits per sample.
// Otherwise, we keep the bits sample as is since we still request fixed
// point PCM. In that case the closest match is typically in float format
// (KSDATAFORMAT_SUBTYPE_IEEE_FLOAT).
if (CoreAudioUtil::WaveFormatWrapper(closest_match.get()).IsPcm()) {
input_format->wBitsPerSample = closest_match->wBitsPerSample;
}
input_format->nBlockAlign =
(input_format->wBitsPerSample / 8) * input_format->nChannels;
input_format->nAvgBytesPerSec =
input_format->nSamplesPerSec * input_format->nBlockAlign;
if (IsSupportedFormatForConversion(&input_format_)) {
SendLogMessage(
"%s => (WARNING: Captured audio will be converted: [%s] ==> [%s])",
__func__, CoreAudioUtil::WaveFormatToString(&input_format_).c_str(),
CoreAudioUtil::WaveFormatToString(&output_format_).c_str());
SetupConverterAndStoreFormatInfo();
// Indicate that we're good to go with a close match.
hresult = S_OK;
}
}
// At this point, |hresult| == S_OK if the desired format is supported. If
// |hresult| == S_FALSE, the OS supports a closest match but we don't support
// conversion to it. Thus, SUCCEEDED() or FAILED() can't be used to determine
// if the desired format is supported.
*hr = hresult;
return (hresult == S_OK);
}
void WASAPIAudioInputStream::SetupConverterAndStoreFormatInfo() {
// Ideally, we want a 1:1 ratio between the buffers we get and the buffers
// we give to OnData so that each buffer we receive from the OS can be
// directly converted to a buffer that matches with what was asked for.
const double buffer_ratio =
output_format_.nSamplesPerSec / static_cast<double>(packet_size_frames_);
double new_frames_per_buffer =
input_format_.Format.nSamplesPerSec / buffer_ratio;
const auto input_layout = GuessChannelLayout(input_format_.Format.nChannels);
DCHECK_NE(CHANNEL_LAYOUT_UNSUPPORTED, input_layout);
const auto output_layout = GuessChannelLayout(output_format_.nChannels);
DCHECK_NE(CHANNEL_LAYOUT_UNSUPPORTED, output_layout);
const AudioParameters input(AudioParameters::AUDIO_PCM_LOW_LATENCY,
input_layout, input_format_.Format.nSamplesPerSec,
static_cast<int>(new_frames_per_buffer));
const AudioParameters output(AudioParameters::AUDIO_PCM_LOW_LATENCY,
output_layout, output_format_.nSamplesPerSec,
packet_size_frames_);
converter_ = std::make_unique<AudioConverter>(input, output, false);
converter_->AddInput(this);
converter_->PrimeWithSilence();
convert_bus_ = AudioBus::Create(output);
// Update our packet size assumptions based on the new format.
const auto new_bytes_per_buffer = static_cast<int>(new_frames_per_buffer) *
input_format_.Format.nBlockAlign;
packet_size_frames_ = new_bytes_per_buffer / input_format_.Format.nBlockAlign;
packet_size_bytes_ = new_bytes_per_buffer;
frame_size_bytes_ = input_format_.Format.nBlockAlign;
imperfect_buffer_size_conversion_ =
std::modf(new_frames_per_buffer, &new_frames_per_buffer) != 0.0;
if (imperfect_buffer_size_conversion_) {
SendLogMessage("%s => (WARNING: Audio capture conversion requires a FIFO)",
__func__);
}
}
HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
DCHECK_EQ(OPEN_RESULT_OK, open_result_);
SendLogMessage("%s()", __func__);
DWORD flags;
// Use event-driven mode only for regular input devices. For loopback the
// EVENTCALLBACK flag is specified when initializing
// |audio_render_client_for_loopback_|.
if (AudioDeviceDescription::IsLoopbackDevice(device_id_)) {
flags = AUDCLNT_STREAMFLAGS_LOOPBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
} else {
flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
}
// Initialize the audio stream between the client and the device.
// We connect indirectly through the audio engine by using shared mode.
// The buffer duration is set to 100 ms, which reduces the risk of glitches.
// It would normally be set to 0 and the minimum buffer size to ensure that
// glitches do not occur would be used (typically around 22 ms). There are
// however cases when there are glitches anyway and it's avoided by setting a
// larger buffer size. The larger size does not create higher latency for
// properly implemented drivers.
HRESULT hr = audio_client_->Initialize(
AUDCLNT_SHAREMODE_SHARED, flags,
100 * 1000 * 10, // Buffer duration, 100 ms expressed in 100-ns units.
0, // Device period, n/a for shared mode.
reinterpret_cast<const WAVEFORMATEX*>(&input_format_),
AudioDeviceDescription::IsCommunicationsDevice(device_id_)
? &kCommunicationsSessionId
: nullptr);
if (FAILED(hr)) {
SendLogMessage("%s => (ERROR: IAudioClient::Initialize=[%s])", __func__,
ErrorToString(hr).c_str());
open_result_ = OPEN_RESULT_AUDIO_CLIENT_INIT_FAILED;
base::UmaHistogramSparse("Media.Audio.Capture.Win.InitError", hr);
MaybeReportFormatRelatedInitError(hr);
return hr;
}
// Retrieve the length of the endpoint buffer shared between the client
// and the audio engine. The buffer length determines the maximum amount
// of capture data that the audio engine can read from the endpoint buffer
// during a single processing pass.
hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_GET_BUFFER_SIZE_FAILED;
return hr;
}
const int endpoint_buffer_size_ms =
static_cast<double>(endpoint_buffer_size_frames_ * 1000) /
input_format_.Format.nSamplesPerSec +
0.5;
SendLogMessage("%s => (endpoint_buffer_size_frames=%u (%d ms))", __func__,
endpoint_buffer_size_frames_, endpoint_buffer_size_ms);
#ifndef NDEBUG
// The period between processing passes by the audio engine is fixed for a
// particular audio endpoint device and represents the smallest processing
// quantum for the audio engine. This period plus the stream latency between
// the buffer and endpoint device represents the minimum possible latency
// that an audio application can achieve.
REFERENCE_TIME device_period_shared_mode = 0;
REFERENCE_TIME device_period_exclusive_mode = 0;
HRESULT hr_dbg = audio_client_->GetDevicePeriod(
&device_period_shared_mode, &device_period_exclusive_mode);
if (SUCCEEDED(hr_dbg)) {
// The 5000 addition is to round end result to closest integer.
const int device_period_ms = (device_period_shared_mode + 5000) / 10000;
DVLOG(1) << "Device period: " << device_period_ms << " ms";
}
REFERENCE_TIME latency = 0;
hr_dbg = audio_client_->GetStreamLatency(&latency);
if (SUCCEEDED(hr_dbg)) {
// The 5000 addition is to round end result to closest integer.
const int latency_ms = (device_period_shared_mode + 5000) / 10000;
DVLOG(1) << "Stream latency: " << latency_ms << " ms";
}
#endif
// Set the event handle that the audio engine will signal each time a buffer
// becomes ready to be processed by the client.
//
// In loopback case the capture device doesn't receive any events, so we
// need to create a separate playback client to get notifications. According
// to MSDN:
//
// A pull-mode capture client does not receive any events when a stream is
// initialized with event-driven buffering and is loopback-enabled. To
// work around this, initialize a render stream in event-driven mode. Each
// time the client receives an event for the render stream, it must signal
// the capture client to run the capture thread that reads the next set of
// samples from the capture endpoint buffer.
//
// http://msdn.microsoft.com/en-us/library/windows/desktop/dd316551(v=vs.85).aspx
if (AudioDeviceDescription::IsLoopbackDevice(device_id_)) {
SendLogMessage("%s => (WARNING: loopback mode is selected)", __func__);
hr = endpoint_device_->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr,
&audio_render_client_for_loopback_);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_LOOPBACK_ACTIVATE_FAILED;
return hr;
}
hr = audio_render_client_for_loopback_->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST, 0, 0,
reinterpret_cast<const WAVEFORMATEX*>(&input_format_),
AudioDeviceDescription::IsCommunicationsDevice(device_id_)
? &kCommunicationsSessionId
: nullptr);
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_LOOPBACK_INIT_FAILED;
return hr;
}
hr = audio_render_client_for_loopback_->SetEventHandle(
audio_samples_ready_event_.Get());
} else {
hr = audio_client_->SetEventHandle(audio_samples_ready_event_.Get());
}
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_SET_EVENT_HANDLE;
return hr;
}
// Get access to the IAudioCaptureClient interface. This interface
// enables us to read input data from the capture endpoint buffer.
hr = audio_client_->GetService(IID_PPV_ARGS(&audio_capture_client_));
if (FAILED(hr)) {
open_result_ = OPEN_RESULT_NO_CAPTURE_CLIENT;
return hr;
}
// Obtain a reference to the ISimpleAudioVolume interface which enables
// us to control the master volume level of an audio session.
hr = audio_client_->GetService(IID_PPV_ARGS(&simple_audio_volume_));
if (FAILED(hr))
open_result_ = OPEN_RESULT_NO_AUDIO_VOLUME;
return hr;
}
void WASAPIAudioInputStream::ReportOpenResult(HRESULT hr) {
DCHECK(!opened_);
UMA_HISTOGRAM_ENUMERATION("Media.Audio.Capture.Win.Open", open_result_,
OPEN_RESULT_MAX + 1);
if (open_result_ != OPEN_RESULT_OK &&
open_result_ != OPEN_RESULT_OK_WITH_RESAMPLING) {
SendLogMessage(
"%s", GetOpenLogString(open_result_, hr, input_format_, output_format_)
.c_str());
}
}
void WASAPIAudioInputStream::MaybeReportFormatRelatedInitError(
HRESULT hr) const {
if (hr != AUDCLNT_E_UNSUPPORTED_FORMAT && hr != E_INVALIDARG)
return;
const FormatRelatedInitError format_related_error =
hr == AUDCLNT_E_UNSUPPORTED_FORMAT
? converter_.get()
? FormatRelatedInitError::kUnsupportedFormatWithFormatConversion
: FormatRelatedInitError::kUnsupportedFormat
// Otherwise |hr| == E_INVALIDARG.
: converter_.get()
? FormatRelatedInitError::kInvalidArgumentWithFormatConversion
: FormatRelatedInitError::kInvalidArgument;
base::UmaHistogramEnumeration(
"Media.Audio.Capture.Win.InitError.FormatRelated", format_related_error,
FormatRelatedInitError::kCount);
}
double WASAPIAudioInputStream::ProvideInput(AudioBus* audio_bus,
uint32_t frames_delayed) {
fifo_->Consume()->CopyTo(audio_bus);
return 1.0;
}
void WASAPIAudioInputStream::UpdateGlitchCount(UINT64 device_position) {
if (expected_next_device_position_ != 0) {
if (device_position > expected_next_device_position_) {
++total_glitches_;
auto lost_frames = device_position - expected_next_device_position_;
total_lost_frames_ += lost_frames;
if (lost_frames > largest_glitch_frames_)
largest_glitch_frames_ = lost_frames;
}
}
}
void WASAPIAudioInputStream::ReportAndResetGlitchStats() {
UMA_HISTOGRAM_COUNTS_1M("Media.Audio.Capture.Glitches", total_glitches_);
double lost_frames_ms =
(total_lost_frames_ * 1000) / input_format_.Format.nSamplesPerSec;
SendLogMessage(
"%s => (total glitches=[%d], total frames lost=[%llu/%.0lf ms])",
__func__, total_glitches_, total_lost_frames_, lost_frames_ms);
if (total_glitches_ != 0) {
UMA_HISTOGRAM_LONG_TIMES("Media.Audio.Capture.LostFramesInMs",
base::Milliseconds(lost_frames_ms));
int64_t largest_glitch_ms =
(largest_glitch_frames_ * 1000) / input_format_.Format.nSamplesPerSec;
UMA_HISTOGRAM_CUSTOM_TIMES("Media.Audio.Capture.LargestGlitchMs",
base::Milliseconds(largest_glitch_ms),
base::Milliseconds(1), base::Minutes(1), 50);
}
// TODO(https://crbug.com/825744): It can be possible to replace
// "Media.Audio.Capture.Glitches" with this new (simplified) metric instead.
base::UmaHistogramCounts1M("Media.Audio.Capture.Win.Glitches",
num_data_discontinuity_warnings_);
SendLogMessage("%s => (discontinuity warnings=[%" PRIu64 "])", __func__,
num_data_discontinuity_warnings_);
SendLogMessage("%s => (timstamp errors=[%" PRIu64 "])", __func__,
num_timestamp_errors_);
if (num_timestamp_errors_ > 0) {
SendLogMessage("%s => (time until first timestamp error=[%" PRId64 " ms])",
__func__,
time_until_first_timestamp_error_.InMilliseconds());
}
expected_next_device_position_ = 0;
total_glitches_ = 0;
total_lost_frames_ = 0;
largest_glitch_frames_ = 0;
num_data_discontinuity_warnings_ = 0;
num_timestamp_errors_ = 0;
}
} // namespace media