blob: c92fedf0e971daf883a2d9c47059b31f39ec89c0 [file] [log] [blame]
/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_device/win/core_audio_output_win.h"
#include <memory>
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
using Microsoft::WRL::ComPtr;
namespace webrtc {
namespace webrtc_win {
CoreAudioOutput::CoreAudioOutput(bool automatic_restart)
: CoreAudioBase(
CoreAudioBase::Direction::kOutput,
automatic_restart,
[this](uint64_t freq) { return OnDataCallback(freq); },
[this](ErrorType err) { return OnErrorCallback(err); }) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
thread_checker_audio_.Detach();
}
CoreAudioOutput::~CoreAudioOutput() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
Terminate();
}
int CoreAudioOutput::Init() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return 0;
}
int CoreAudioOutput::Terminate() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
StopPlayout();
return 0;
}
int CoreAudioOutput::NumDevices() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
return core_audio_utility::NumberOfActiveDevices(eRender);
}
int CoreAudioOutput::SetDevice(int index) {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_GE(index, 0);
RTC_DCHECK_RUN_ON(&thread_checker_);
return CoreAudioBase::SetDevice(index);
}
int CoreAudioOutput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": "
<< ((device == AudioDeviceModule::kDefaultDevice)
? "Default"
: "DefaultCommunication");
RTC_DCHECK_RUN_ON(&thread_checker_);
return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
}
int CoreAudioOutput::DeviceName(int index,
std::string* name,
std::string* guid) {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(name);
return CoreAudioBase::DeviceName(index, name, guid);
}
void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
audio_device_buffer_ = audio_buffer;
}
bool CoreAudioOutput::PlayoutIsInitialized() const {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return initialized_;
}
int CoreAudioOutput::InitPlayout() {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
RTC_DCHECK(!initialized_);
RTC_DCHECK(!Playing());
RTC_DCHECK(!audio_render_client_);
// Creates an IAudioClient instance and stores the valid interface pointer in
// `audio_client3_`, `audio_client2_`, or `audio_client_` depending on
// platform support. The base class will use optimal output parameters and do
// an event driven shared mode initialization. The utilized format will be
// stored in `format_` and can be used for configuration and allocation of
// audio buffers.
if (!CoreAudioBase::Init()) {
return -1;
}
RTC_DCHECK(audio_client_);
// Configure the playout side of the audio device buffer using `format_`
// after a trivial sanity check of the format structure.
RTC_DCHECK(audio_device_buffer_);
WAVEFORMATEX* format = &format_.Format;
RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
audio_device_buffer_->SetPlayoutSampleRate(format->nSamplesPerSec);
audio_device_buffer_->SetPlayoutChannels(format->nChannels);
// Create a modified audio buffer class which allows us to ask for any number
// of samples (and not only multiple of 10ms) to match the optimal
// buffer size per callback used by Core Audio.
// TODO(henrika): can we share one FineAudioBuffer with the input side?
fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
// Create an IAudioRenderClient for an initialized IAudioClient.
// The IAudioRenderClient interface enables us to write output data to
// a rendering endpoint buffer.
ComPtr<IAudioRenderClient> audio_render_client =
core_audio_utility::CreateRenderClient(audio_client_.Get());
if (!audio_render_client.Get()) {
return -1;
}
ComPtr<IAudioClock> audio_clock =
core_audio_utility::CreateAudioClock(audio_client_.Get());
if (!audio_clock.Get()) {
return -1;
}
// Store valid COM interfaces.
audio_render_client_ = audio_render_client;
audio_clock_ = audio_clock;
initialized_ = true;
return 0;
}
int CoreAudioOutput::StartPlayout() {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
RTC_DCHECK(!Playing());
RTC_DCHECK(fine_audio_buffer_);
RTC_DCHECK(audio_device_buffer_);
if (!initialized_) {
RTC_DLOG(LS_WARNING)
<< "Playout can not start since InitPlayout must succeed first";
}
fine_audio_buffer_->ResetPlayout();
if (!IsRestarting()) {
audio_device_buffer_->StartPlayout();
}
if (!core_audio_utility::FillRenderEndpointBufferWithSilence(
audio_client_.Get(), audio_render_client_.Get())) {
RTC_LOG(LS_WARNING) << "Failed to prepare output endpoint with silence";
}
num_frames_written_ = endpoint_buffer_size_frames_;
if (!Start()) {
return -1;
}
is_active_ = true;
return 0;
}
int CoreAudioOutput::StopPlayout() {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
if (!initialized_) {
return 0;
}
// Release resources allocated in InitPlayout() and then return if this
// method is called without any active output audio.
if (!Playing()) {
RTC_DLOG(LS_WARNING) << "No output stream is active";
ReleaseCOMObjects();
initialized_ = false;
return 0;
}
if (!Stop()) {
RTC_LOG(LS_ERROR) << "StopPlayout failed";
return -1;
}
if (!IsRestarting()) {
RTC_DCHECK(audio_device_buffer_);
audio_device_buffer_->StopPlayout();
}
// Release all allocated resources to allow for a restart without
// intermediate destruction.
ReleaseCOMObjects();
initialized_ = false;
is_active_ = false;
return 0;
}
bool CoreAudioOutput::Playing() {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << is_active_;
return is_active_;
}
// TODO(henrika): finalize support of audio session volume control. As is, we
// are not compatible with the old ADM implementation since it allows accessing
// the volume control with any active audio output stream.
int CoreAudioOutput::VolumeIsAvailable(bool* available) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return IsVolumeControlAvailable(available) ? 0 : -1;
}
// Triggers the restart sequence. Only used for testing purposes to emulate
// a real event where e.g. an active output device is removed.
int CoreAudioOutput::RestartPlayout() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!Playing()) {
return 0;
}
if (!Restart()) {
RTC_LOG(LS_ERROR) << "RestartPlayout failed";
return -1;
}
return 0;
}
bool CoreAudioOutput::Restarting() const {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return IsRestarting();
}
int CoreAudioOutput::SetSampleRate(uint32_t sample_rate) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
sample_rate_ = sample_rate;
return 0;
}
void CoreAudioOutput::ReleaseCOMObjects() {
RTC_DLOG(LS_INFO) << __FUNCTION__;
CoreAudioBase::ReleaseCOMObjects();
if (audio_render_client_.Get()) {
audio_render_client_.Reset();
}
}
bool CoreAudioOutput::OnErrorCallback(ErrorType error) {
RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << as_integer(error);
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
if (!initialized_ || !Playing()) {
return true;
}
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
} else {
RTC_DLOG(LS_WARNING) << "Unsupported error type";
}
return true;
}
bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
if (num_data_callbacks_ == 0) {
RTC_LOG(LS_INFO) << "--- Output audio stream is alive ---";
}
// Get the padding value which indicates the amount of valid unread data that
// the endpoint buffer currently contains.
UINT32 num_unread_frames = 0;
_com_error error = audio_client_->GetCurrentPadding(&num_unread_frames);
if (error.Error() == AUDCLNT_E_DEVICE_INVALIDATED) {
// Avoid breaking the thread loop implicitly by returning false and return
// true instead for AUDCLNT_E_DEVICE_INVALIDATED even it is a valid error
// message. We will use notifications about device changes instead to stop
// data callbacks and attempt to restart streaming .
RTC_DLOG(LS_ERROR) << "AUDCLNT_E_DEVICE_INVALIDATED";
return true;
}
if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
<< core_audio_utility::ErrorToString(error);
return false;
}
// 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. I.e., it is the maximum buffer size we can request when
// calling IAudioRenderClient::GetBuffer().
UINT32 num_requested_frames =
endpoint_buffer_size_frames_ - num_unread_frames;
if (num_requested_frames == 0) {
RTC_DLOG(LS_WARNING)
<< "Audio thread is signaled but no new audio samples are needed";
return true;
}
// Request all available space in the rendering endpoint buffer into which the
// client can later write an audio packet.
uint8_t* audio_data;
error = audio_render_client_->GetBuffer(num_requested_frames, &audio_data);
if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
<< core_audio_utility::ErrorToString(error);
return false;
}
// Update output delay estimate but only about once per second to save
// resources. The estimate is usually stable.
if (num_data_callbacks_ % 100 == 0) {
// TODO(henrika): note that FineAudioBuffer adds latency as well.
latency_ms_ = EstimateOutputLatencyMillis(device_frequency);
if (num_data_callbacks_ % 500 == 0) {
RTC_DLOG(LS_INFO) << "latency: " << latency_ms_;
}
}
// Get audio data from WebRTC and write it to the allocated buffer in
// `audio_data`. The playout latency is not updated for each callback.
fine_audio_buffer_->GetPlayoutData(
rtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data),
num_requested_frames * format_.Format.nChannels),
latency_ms_);
// Release the buffer space acquired in IAudioRenderClient::GetBuffer.
error = audio_render_client_->ReleaseBuffer(num_requested_frames, 0);
if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
<< core_audio_utility::ErrorToString(error);
return false;
}
num_frames_written_ += num_requested_frames;
++num_data_callbacks_;
return true;
}
// TODO(henrika): IAudioClock2::GetDevicePosition could perhaps be used here
// instead. Tried it once, but it crashed for capture devices.
int CoreAudioOutput::EstimateOutputLatencyMillis(uint64_t device_frequency) {
UINT64 position = 0;
UINT64 qpc_position = 0;
int delay_ms = 0;
// Get the device position through output parameter `position`. This is the
// stream position of the sample that is currently playing through the
// speakers.
_com_error error = audio_clock_->GetPosition(&position, &qpc_position);
if (error.Error() == S_OK) {
// Number of frames already played out through the speaker.
const uint64_t num_played_out_frames =
format_.Format.nSamplesPerSec * position / device_frequency;
// Number of frames that have been written to the buffer but not yet
// played out corresponding to the estimated latency measured in number
// of audio frames.
const uint64_t delay_frames = num_frames_written_ - num_played_out_frames;
// Convert latency in number of frames into milliseconds.
webrtc::TimeDelta delay =
webrtc::TimeDelta::Micros(delay_frames * rtc::kNumMicrosecsPerSec /
format_.Format.nSamplesPerSec);
delay_ms = delay.ms();
}
return delay_ms;
}
// Called from OnErrorCallback() when error type is kStreamDisconnected.
// Note that this method is called on the audio thread and the internal restart
// sequence is also executed on that same thread. The audio thread is therefore
// not stopped during restart. Such a scheme also makes the restart process less
// complex.
// Note that, none of the called methods are thread checked since they can also
// be called on the main thread. Thread checkers are instead added on one layer
// above (in audio_device_module.cc) which ensures that the public API is thread
// safe.
// TODO(henrika): add more details.
bool CoreAudioOutput::HandleStreamDisconnected() {
RTC_DLOG(LS_INFO) << "<<<--- " << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
RTC_DCHECK(automatic_restart());
if (StopPlayout() != 0) {
return false;
}
if (!SwitchDeviceIfNeeded()) {
return false;
}
if (InitPlayout() != 0) {
return false;
}
if (StartPlayout() != 0) {
return false;
}
RTC_DLOG(LS_INFO) << __FUNCTION__ << " --->>>";
return true;
}
} // namespace webrtc_win
} // namespace webrtc