1 files changed, 422 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc
new file mode 100644
index 0000000000..c92fedf0e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc
@@ -0,0 +1,422 @@
+/*
+ *  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