Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_output_win.cc

+// Copyright (c) 2011 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 "base/logging.h"
+#include "base/memory/scoped_ptr.h"
+#include "base/utf_string_conversions.h"
+#include "media/audio/audio_util.h"
+#include "media/audio/win/audio_manager_win.h"
+#include "media/audio/win/avrt_wrapper_win.h"
+
+using base::win::ScopedComPtr;
+using base::win::ScopedCOMInitializer;
+
+WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
+                                                 const AudioParameters& params,
+                                                 ERole device_role)
+    : com_init_(ScopedCOMInitializer::kMTA),
+      creating_thread_id_(base::PlatformThread::CurrentId()),
+      manager_(manager),
+      render_thread_(NULL),
+      opened_(false),
+      started_(false),
+      volume_(1.0),
+      endpoint_buffer_size_frames_(0),
+      device_role_(device_role),
+      num_written_frames_(0),
+      source_(NULL) {
+  CHECK(com_init_.succeeded());
+  DCHECK(manager_);
+
+  // 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.
+  format_.nSamplesPerSec = params.sample_rate;
+  format_.wFormatTag = WAVE_FORMAT_PCM;
+  format_.wBitsPerSample = params.bits_per_sample;
+  format_.nChannels = params.channels;
+  format_.nBlockAlign = (format_.wBitsPerSample / 8) * format_.nChannels;
+  format_.nAvgBytesPerSec = format_.nSamplesPerSec * format_.nBlockAlign;
+  format_.cbSize = 0;
+
+  // Size in bytes of each audio frame.
+  frame_size_ = format_.nBlockAlign;
+
+  // 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.GetPacketSize() / format_.nBlockAlign;
+  packet_size_bytes_ = params.GetPacketSize();
+  packet_size_ms_ = (1000.0 * packet_size_frames_) / params.sample_rate;
+  DVLOG(1) << "Number of bytes per audio frame  : " << frame_size_;
+  DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
+  DVLOG(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() {
+  DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
+  if (opened_)
+    return true;
+
+  // Obtain a reference to the IMMDevice interface of the default rendering
+  // device with the specified role.
+  HRESULT hr = SetRenderDevice(device_role_);
+  if (FAILED(hr)) {
+    HandleError(hr);
+    return false;
+  }
+
+  // Obtain an IAudioClient interface which enables us to create and initialize
+  // an audio stream between an audio application and the audio engine.
+  hr = ActivateRenderDevice();
+  if (FAILED(hr)) {
+    HandleError(hr);
+    return false;
+  }
+
+  // Retrieve the stream format which the audio engine uses for its internal
+  // processing/mixing of shared-mode streams.
+  hr = GetAudioEngineStreamFormat();
+  if (FAILED(hr)) {
+    HandleError(hr);
+    return false;
+  }
+
+  // Verify that the selected audio endpoint supports the specified format
+  // set during construction.
+  if (!DesiredFormatIsSupported()) {
+    hr = E_INVALIDARG;
+    HandleError(hr);
+    return false;
+  }
+
+  // Initialize the audio stream between the client and the device using
+  // shared mode and a lowest possible glitch-free latency.
+  hr = InitializeAudioEngine();
+  if (FAILED(hr)) {
+    HandleError(hr);
+    return false;
+  }
+
+  opened_ = true;
+
+  return true;
+}
+
+void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
+  DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
+  DCHECK(callback);
+  DCHECK(opened_);
+
+  if (!opened_)
+    return;
+
+  if (started_)
+    return;
+
+  source_ = callback;
+
+  // Avoid start-up glitches by filling up the endpoint buffer with "silence"
+  // before starting the stream.
+  BYTE* data_ptr = NULL;
+  HRESULT hr = audio_render_client_->GetBuffer(endpoint_buffer_size_frames_,
+                                               &data_ptr);
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Failed to use rendering audio buffer: " << std::hex << hr;
+    return;
+  }
+
+  // Using the AUDCLNT_BUFFERFLAGS_SILENT flag eliminates the need to
+  // explicitly write silence data to the rendering buffer.
+  audio_render_client_->ReleaseBuffer(endpoint_buffer_size_frames_,
+                                      AUDCLNT_BUFFERFLAGS_SILENT);
+  num_written_frames_ = endpoint_buffer_size_frames_;
+
+  // Sanity check: verify that the endpoint buffer is filled with silence.
+  UINT32 num_queued_frames = 0;
+  audio_client_->GetCurrentPadding(&num_queued_frames);
+  DCHECK(num_queued_frames == num_written_frames_);
+
+  // Create and start the thread that will drive the rendering by waiting for
+  // render events.
+  render_thread_ = new base::DelegateSimpleThread(this, "wasapi_render_thread");
+  render_thread_->Start();
+  if (!render_thread_->HasBeenStarted()) {
+    DLOG(ERROR) << "Failed to start WASAPI render thread.";
+    return;
+  }
+
+  // Start streaming data between the endpoint buffer and the audio engine.
+  hr = audio_client_->Start();
+  if (FAILED(hr)) {
+    SetEvent(stop_render_event_.Get());
+    render_thread_->Join();
+    render_thread_ = NULL;
+    HandleError(hr);
+    return;
+  }
+
+  started_ = true;
+}
+
+void WASAPIAudioOutputStream::Stop() {
+  DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
+  if (!started_)
+    return;
+
+  // Shut down the render thread.
+  if (stop_render_event_.IsValid()) {
+    SetEvent(stop_render_event_.Get());
+  }
+
+  // Stop output audio streaming.
+  HRESULT hr = audio_client_->Stop();
+  DLOG_IF(ERROR, FAILED(hr)) << "Failed to stop output streaming: "
+                             << std::hex << hr;
+
+  // Wait until the thread completes and perform cleanup.
+  if (render_thread_) {
+    SetEvent(stop_render_event_.Get());
+    render_thread_->Join();
+    render_thread_ = NULL;
+  }
+
+  started_ = false;
+}
+
+void 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) {
+  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) {
+  *volume = static_cast<double>(volume_);
+}
+
+// static
+double WASAPIAudioOutputStream::HardwareSampleRate(ERole device_role) {
+  // It is assumed that this static method is called from a COM thread, i.e.,
+  // CoInitializeEx() is not called here again to avoid STA/MTA conflicts.
+  ScopedComPtr<IMMDeviceEnumerator> enumerator;
+  HRESULT hr =  CoCreateInstance(__uuidof(MMDeviceEnumerator),
+                                 NULL,
+                                 CLSCTX_INPROC_SERVER,
+                                 __uuidof(IMMDeviceEnumerator),
+                                 enumerator.ReceiveVoid());
+  if (FAILED(hr)) {
+    NOTREACHED() << "error code: " << std::hex << hr;
+    return 0.0;
+  }
+
+  ScopedComPtr<IMMDevice> endpoint_device;
+  hr = enumerator->GetDefaultAudioEndpoint(eRender,
+                                           device_role,
+                                           endpoint_device.Receive());
+  if (FAILED(hr)) {
+    // This will happen if there's no audio output device found or available
+    // (e.g. some audio cards that have outputs will still report them as
+    // "not found" when no speaker is plugged into the output jack).
+    LOG(WARNING) << "No audio end point: " << std::hex << hr;
+    return 0.0;
+  }
+
+  ScopedComPtr<IAudioClient> audio_client;
+  hr = endpoint_device->Activate(__uuidof(IAudioClient),
+                                 CLSCTX_INPROC_SERVER,
+                                 NULL,
+                                 audio_client.ReceiveVoid());
+  if (FAILED(hr)) {
+    NOTREACHED() << "error code: " << std::hex << hr;
+    return 0.0;
+  }
+
+  base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format;
+  hr = audio_client->GetMixFormat(&audio_engine_mix_format);
+  if (FAILED(hr)) {
+    NOTREACHED() << "error code: " << std::hex << hr;
+    return 0.0;
+  }
+
+  return static_cast<double>(audio_engine_mix_format->nSamplesPerSec);
+}
+
+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[2] = {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;
+
+  // Render audio until stop event or error.
+  while (playing && !error) {
+    // Wait for a close-down event or a new render event.
+    DWORD wait_result = WaitForMultipleObjects(2, 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.
+          UINT32 num_queued_frames = 0;
+          uint8* audio_data = NULL;
+
+          // 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);
+
+          // Determine 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 =
+              endpoint_buffer_size_frames_ - num_queued_frames;
+
+          // Check if there is enough available space to fit the packet size
+          // specified by the client.
+          if (FAILED(hr) || (num_available_frames < packet_size_frames_))
+            continue;
+
+          // Derive the number of packets we need get from the client to
+          // fill up the available area in the endpoint buffer.
+          size_t num_packets = (num_available_frames / packet_size_frames_);
+
+          // Get data from the client/source.
+          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;
+              continue;
+            }
+
+            // 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_.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) *  frame_size_;
+            }
+
+            // 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.
+            uint32 num_filled_bytes = source_->OnMoreData(
+                this, audio_data, packet_size_bytes_,
+                AudioBuffersState(0, audio_delay_bytes));
+
+            // Perform in-place, software-volume adjustments.
+            media::AdjustVolume(audio_data,
+                                num_filled_bytes,
+                                format_.nChannels,
+                                format_.wBitsPerSample >> 3,
+                                volume_);
+
+            // Zero out the part of the packet which has not been filled by
+            // the client. Using silence is the least bad option in this
+            // situation.
+            if (num_filled_bytes < packet_size_bytes_) {
+              memset(&audio_data[num_filled_bytes], 0,
+                    (packet_size_bytes_ - num_filled_bytes));
+            }
+
+            // Release the buffer space acquired in the GetBuffer() call.
+            DWORD flags = 0;
+            audio_render_client_->ReleaseBuffer(packet_size_frames_,
+                                                flags);
+
+            num_written_frames_ += packet_size_frames_;
+          }
+        }
+        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";
+  }
+}
+
+void WASAPIAudioOutputStream::HandleError(HRESULT err) {
+  NOTREACHED() << "Error code: " << std::hex << err;
+  if (source_)
+    source_->OnError(this, static_cast<int>(err));
+}
+
+HRESULT WASAPIAudioOutputStream::SetRenderDevice(ERole device_role) {
+  ScopedComPtr<IMMDeviceEnumerator> enumerator;
+  HRESULT hr =  CoCreateInstance(__uuidof(MMDeviceEnumerator),
+                                 NULL,
+                                 CLSCTX_INPROC_SERVER,
+                                 __uuidof(IMMDeviceEnumerator),
+                                 enumerator.ReceiveVoid());
+  if (SUCCEEDED(hr)) {
+    // Retrieve the default render audio endpoint for the specified role.
+    // Note that, in Windows Vista, the MMDevice API supports device roles
+    // but the system-supplied user interface programs do not.
+    hr = enumerator->GetDefaultAudioEndpoint(eRender,
+                                             device_role,
+                                             endpoint_device_.Receive());
+
+    // Verify that the audio endpoint device is active. That is, the audio
+    // adapter that connects to the endpoint device is present and enabled.
+    DWORD state = DEVICE_STATE_DISABLED;
+    hr = endpoint_device_->GetState(&state);
+    if (SUCCEEDED(hr)) {
+      if (!(state & DEVICE_STATE_ACTIVE)) {
+        DLOG(ERROR) << "Selected render device is not active.";
+        hr = E_ACCESSDENIED;
+      }
+    }
+  }
+
+  return hr;
+}
+
+HRESULT WASAPIAudioOutputStream::ActivateRenderDevice() {
+  // Creates and activates an IAudioClient COM object given the selected
+  // render endpoint device.
+  HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient),
+                                          CLSCTX_INPROC_SERVER,
+                                          NULL,
+                                          audio_client_.ReceiveVoid());
+  return hr;
+}
+
+HRESULT WASAPIAudioOutputStream::GetAudioEngineStreamFormat() {
+  // Retrieve the stream format that the audio engine uses for its internal
+  // processing/mixing of shared-mode streams.
+  return audio_client_->GetMixFormat(&audio_engine_mix_format_);
+}
+
+bool WASAPIAudioOutputStream::DesiredFormatIsSupported() {
+  // In shared mode, the audio engine always supports the mix format,
+  // which is stored in the |audio_engine_mix_format_| member. In addition,
+  // the audio engine *might* support similar formats that have the same
+  // sample rate and number of channels as the mix format but differ in
+  // the representation of audio sample values.
+  base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
+  HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
+                                                &format_,
+                                                &closest_match);
+  DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
+                                << "but a closest match exists.";
+  return (hr == S_OK);
+}
+
+HRESULT WASAPIAudioOutputStream::InitializeAudioEngine() {
+  // TODO(henrika): this buffer scheme is still under development.
+  // The exact details are yet to be determined based on tests with different
+  // audio clients.
+  int glitch_free_buffer_size_ms = static_cast<int>(packet_size_ms_ + 0.5);
+  if (audio_engine_mix_format_->nSamplesPerSec == 48000) {
+    // Initial tests have shown that we have to add 10 ms extra to
+    // ensure that we don't run empty for any packet size.
+    glitch_free_buffer_size_ms += 10;
+  } else if (audio_engine_mix_format_->nSamplesPerSec == 44100) {
+    // Initial tests have shown that we have to add 20 ms extra to
+    // ensure that we don't run empty for any packet size.
+    glitch_free_buffer_size_ms += 20;
+  } else {
+    glitch_free_buffer_size_ms += 20;
+  }
+  DVLOG(1) << "glitch_free_buffer_size_ms: " << glitch_free_buffer_size_ms;
+  REFERENCE_TIME requested_buffer_duration_hns =
+      static_cast<REFERENCE_TIME>(glitch_free_buffer_size_ms * 10000);
+
+  // Initialize the audio stream between the client and the device.
+  // We connect indirectly through the audio engine by using shared mode
+  // and WASAPI is initialized in an event driven mode.
+  // Note that this API ensures that the buffer is never smaller than the
+  // minimum buffer size needed to ensure glitch-free rendering.
+  // If we requests a buffer size that is smaller than the audio engine's
+  // minimum required buffer size, the method sets the buffer size to this
+  // minimum buffer size rather than to the buffer size requested.
+  HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
+                                         AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                         AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                         requested_buffer_duration_hns,
+                                         0,
+                                         &format_,
+                                         NULL);
+  if (FAILED(hr))
+    return hr;
+
+  // Retrieve the length of the endpoint buffer shared between the client
+  // and the audio engine. The buffer length the buffer length determines
+  // the maximum amount of rendering data that the client can write to
+  // the endpoint buffer during a single processing pass.
+  // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+  hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
+  if (FAILED(hr))
+    return hr;
+  DVLOG(1) << "endpoint buffer size: " << endpoint_buffer_size_frames_
+           << " [frames]";
+#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 in shared mode.
+  REFERENCE_TIME default_device_period = 0;
+  REFERENCE_TIME minimum_device_period = 0;
+  HRESULT hr_dbg = audio_client_->GetDevicePeriod(&default_device_period,
+                                                  &minimum_device_period);
+  if (SUCCEEDED(hr_dbg)) {
+    // Shared mode device period.
+    DVLOG(1) << "default device period: "
+             << static_cast<double>(default_device_period / 10000.0)
+             << " [ms]";
+    // Exclusive mode device period.
+    DVLOG(1) << "minimum device period: "
+             << static_cast<double>(minimum_device_period / 10000.0)
+             << " [ms]";
+  }
+
+  REFERENCE_TIME latency = 0;
+  hr_dbg = audio_client_->GetStreamLatency(&latency);
+  if (SUCCEEDED(hr_dbg)) {
+    DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0)
+             << " [ms]";
+  }
+#endif
+
+  // Set the event handle that the audio engine will signal each time
+  // a buffer becomes ready to be processed by the client.
+  hr = audio_client_->SetEventHandle(audio_samples_render_event_.Get());
+  if (FAILED(hr))
+    return hr;
+
+  // Get access to the IAudioRenderClient interface. This interface
+  // enables us to write output data to a rendering endpoint buffer.
+  // The methods in this interface manage the movement of data packets
+  // that contain audio-rendering data.
+  hr = audio_client_->GetService(__uuidof(IAudioRenderClient),
+                                 audio_render_client_.ReceiveVoid());
+  return hr;
+}