Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_output_win.cc

Index: media/audio/win/audio_low_latency_output_win.cc
===================================================================
--- media/audio/win/audio_low_latency_output_win.cc	(revision 0)
+++ media/audio/win/audio_low_latency_output_win.cc	(revision 0)
@@ -0,0 +1,571 @@
+// 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),

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

do you want to add a check to the constructor (as is indicated in a comment in
the header file):

CHECK(com_init_.succeeded());

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+      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) {
+  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";

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

nit: "Failed to load avrt.dll"

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+
+  // 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;

[Raymond Toy (Google), 2011/11/03 22:49:56]

Is it possible for GetPacketSize() to return something that is not a multiple of
nBlockAlign?  If so, do we care if packet_size_frames is rounded down?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

The result will always be equal to params.samples_per_packet. I could have used
it but felt that it was more clear to show what was meant by and audio frame.
I can change if you don't like he current proposal.

[Raymond Toy (Google), 2011/11/04 16:58:02]

No, this is fine.

+  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() {

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

ponder: Should we DCHECK in these methods (Open, Start, Stop, Close) that the
calling thread is the construction thread?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Good point. Makes sense here since we need a certain COM environment.

Added DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);

+  // Verify that we are not already opened.
+  if (opened_)
+    return false;
+
+  // 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(callback);
+  DCHECK(opened_);
+
+  if (!opened_)
+    return;
+
+  if (started_)
+    return;
+
+  source_ = callback;
+
+  // 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();

[Niklas Enbom, 2011/11/03 15:32:09]

Why start the thread already here? What if following initialization calls fails?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

Good point, will modify. 

+
+  // 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 (SUCCEEDED(hr)) {
+    // 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_);
+  }
+
+  // Start streaming data between the endpoint buffer and the audio engine.
+  hr = audio_client_->Start();

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

If this fails, should we call HandleError()?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Improved:

 if (FAILED(hr)) {
    SetEvent(stop_render_event_.Get());
    render_thread_->Join();
    render_thread_ = NULL;
    HandleError(hr);
    return;
  }

+  DLOG_IF(ERROR, FAILED(hr)) << "Failed to start output streaming: "
+                             << std::hex << hr;
+
+  started_ = SUCCEEDED(hr);
+}
+
+void WASAPIAudioOutputStream::Stop() {
+  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() {
+  // 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) {
+  if (volume < 0.0f || volume > 1.0f)
+    return;

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

is this something we expect to happen?  If this should never happen, add a
NOTREACHED().

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+  volume_ = static_cast<float>(volume);

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

nit: casting from double to float isn't exact, so you might want to cast first
to a temporary, do the |if| check, then assign from the temporary to volume_.

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Fixed ;-)

+}
+
+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;
+  }

[Raymond Toy (Google), 2011/11/03 22:49:56]

Should we return 0.0 here, like we do in other NOTREACHED places?  If not,
perhaps a comment saying why we don't would be appropriate.

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

Fixed.

+
+  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);

[Raymond Toy (Google), 2011/11/03 22:49:56]

It's a little confusing here that hr can be the result of GetServices or
GetFrequency.  Does it matter which one failed or does both GetServices and
GetFrequency need to succeed for iAudioClock to have succeeded?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

If GetService() fails, error is true and the thread loop is never entered.
If GetServics() is OK but GetFrequency() fails error is true as well.
Hence, both calls must succeed to enable playout. If any of them fails, we will
not be able to estimate the delay.

+  PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: "
+                        << std::hex << hr;
+
+  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 (num_available_frames < packet_size_frames_) {

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

nit: remove {}

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+            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++) {

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

++n

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+            // 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);
+
+            // 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.
+            if (SUCCEEDED(hr) && audio_data) {

[Raymond Toy (Google), 2011/11/03 22:49:56]

What happens if GetBuffer doesn't succeed?  Is there anything that needs to be
done?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

It is an extremely rare event and I have not been able to provoke it even if all
devices are removed. It can fail if there is a mismatch in buffer size between
GetBuffer and ReleaseBuffer() but given that it all works that can not happen.

Anyhow, I'll see if I can add an additional DVLOG(1).

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

if it does fail though, should error be set to true to avoid a potential
infinite loop?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+              UINT64 position = 0;
+              int audio_delay_bytes = 0;
+              hr = audio_clock->GetPosition(&position, NULL);
+              if (SUCCEEDED(hr)) {

[Raymond Toy (Google), 2011/11/03 22:49:56]

Same thing.  What happens if GetPosition doesn't succed?  Is there anything that
needs to done?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

The user will receive 0 as audio delay and can see that something is wrong that
way. I might extend logging here as well.

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

and set error to true?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

I don't want to stop rendering even if delay estimation fails. Providing 0 as
estimate feels sufficient.

+                // 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.
+              if (num_filled_bytes < packet_size_bytes_) {
+                memset(&audio_data[num_filled_bytes],
+                       0,

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

move to the line above

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+                      (packet_size_bytes_ - num_filled_bytes));

[Niklas Enbom, 2011/11/03 15:46:47]

Why, since silence is better than repeat in case of a glitch?

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

Do you mean that you would like me to add better comment?

[henrika (OOO until Aug 14), 2011/11/07 12:37:40]

Modified the comment.

+              }
+            }
+
+            // Release the buffer space acquired in the GetBuffer() call.
+            DWORD flags(0);

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

flags = 0

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+            hr = audio_render_client_->ReleaseBuffer(packet_size_frames_,
+                                                     flags);
+
+            num_written_frames_ += packet_size_frames_;
+          }
+        }
+        break;
+      default:
+        error = true;
+        break;
+    }
+  }
+
+  if (playing && error) {
+    // TODO(henrika): perhaps it worth improving the cleanup here by e.g.
+    // stopping the audio client, joining the thread etc.?
+    NOTREACHED() << "WASAPI rendering failed with error code "

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

PLOG(ERROR)?
NOTREACHED is a debug-only assert for code that we never expect to run.  It
looks like this could potentially hit and getting a log in release mode might be
useful.

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Thanks. Improved this section.

+                 << GetLastError();
+  }
+
+  // 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;

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

Assert called on correct thread?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Not needed. Called from Open(), DCHECK() is done there instead.

+  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;
+      }
+    }
+  }

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

else log error + hr?

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

The called does this as a result of hr. Should be sufficient.

+
+  return hr;

[Raymond Toy (Google), 2011/11/03 22:49:56]

hr can contain the error code from up to 3 different functions.  Will the caller
of SetRenderDevice ever need to know what actually failed?  Or is failure
enough?

[henrika (OOO until Aug 14), 2011/11/07 12:37:40]

The Open() call consist of five sub functions where SetRenderDevice() is one of
them. I feel that it is enough granularity to return "true" or "false" here.

+}
+
+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);

[tommi (sloooow) - chröme, 2011/11/07 11:47:03]

indent

[henrika (OOO until Aug 14), 2011/11/08 12:18:37]

Done.

+
+  // 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.
+  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)) {
+    DVLOG(1) << "device period: "
+             << static_cast<double>(device_period_shared_mode / 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;

[Raymond Toy (Google), 2011/11/03 22:49:56]

As above, hr can be the result of several different failed function calls.  Does
the actual value matter to the caller of InitializeAudioEngine need to know or
is failure enough?

[henrika (OOO until Aug 14), 2011/11/07 12:37:40]

Failure is enough. The HRESULT code will contain the essential information.
Keeping it as is. Hope that is OK.

+}
Property changes on: media\audio\win\audio_low_latency_output_win.cc
___________________________________________________________________
Added: svn:eol-style
   + LF