Low-latency AudioInputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_input_win.cc

Index: media/audio/win/audio_low_latency_input_win.cc
===================================================================
--- media/audio/win/audio_low_latency_input_win.cc	(revision 0)
+++ media/audio/win/audio_low_latency_input_win.cc	(revision 0)
@@ -0,0 +1,523 @@
+// 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_input_win.h"
+
+#include <comdef.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"
+
+using base::win::ScopedComPtr;
+
+#ifndef NDEBUG
+static void DLogFormat(const char* str, const WAVEFORMATEX* format) {
+  DLOG(INFO) << str << std::endl

[scherkus (not reviewing), 2011/10/18 21:12:30]

as per http://dev.chromium.org/developers/coding-style#TOC-Logging INFO should
rarely be used -- a lot of this looks like debugging only information so please
use DVLOG(1) instead

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Good idea. Done.

PS, the style guide states "Do not use LOG(INFO)", I use DLOG().

+             << " wFormatTag     : " << format->wFormatTag << std::endl
+             << " nChannels      : " << format->nChannels << std::endl
+             << " nSamplesPerSec : " << format->nSamplesPerSec << std::endl
+             << " nAvgBytesPerSec: " << format->nAvgBytesPerSec << std::endl
+             << " wBitsPerSample : " << format->wBitsPerSample << std::endl
+             << " nBlockAlign    : " << format->nBlockAlign << std::endl
+             << " cbSize         : " << format->cbSize << std::endl;
+}
+#endif
+
+WASAPIAudioInputStream::WASAPIAudioInputStream(
+    AudioManagerWin* manager, const AudioParameters& params, ERole device_role)
+    : manager_(manager),
+      capture_thread_(NULL),
+      opened_(false),
+      started_(false),
+      endpoint_buffer_size_frames_(0),
+      device_role_(device_role),
+      sink_(NULL) {
+  DCHECK(manager_);
+
+  // Load the Avrt DLL if not already loaded. Required to support MMCSS.
+  DCHECK(avrt_.Initialize());

[scherkus (not reviewing), 2011/10/18 21:12:30]

what should we do if this fails?

in release mode we'll continue executing

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Yes and that is OK since all will work fine anyhow (simply no fancy priority
boost). A LOG(WARNING) << "Failed to enable MMCSS" will also be added as soon as
capturing starts also in Release() mode.

Extended message at Initialize() as well.

[tommi (sloooow) - chröme, 2011/10/19 20:15:37]

Never use DCHECK for things that need to exist in release builds.  In this case,
this call to Initialize() won't be there.

+
+  // Set up the desired capture 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;
+#ifndef NDEBUG

[scherkus (not reviewing), 2011/10/18 21:12:30]

is this really needed?

DLOG() in release mode compiles to zero code (see base/logging.h) so I'd assume
a smart compiler would also recognize that a static empty method is also not
worth produce code

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

I removed this extra logging.

+  DLogFormat("Desired capture format:", &format_);
+#endif
+
+  // Size in bytes of each audio frame.
+  frame_size_ = format_.nBlockAlign;
+  // Store size of audio packets which we expect to get from the audio
+  // endpoint device in each capture event.
+  packet_size_frames_ = params.GetPacketSize() / format_.nBlockAlign;
+  packet_size_bytes_ = params.GetPacketSize();
+  DLOG(INFO) << "Number of bytes per audio frame  : " << frame_size_;
+  DLOG(INFO) << "Number of audio frames per packet: " << packet_size_frames_;
+
+  // 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());
+
+  ms_to_frame_count_ = static_cast<double>(params.sample_rate) / 1000.0;
+
+  LARGE_INTEGER performance_frequency;
+  if (QueryPerformanceFrequency(&performance_frequency)) {
+    perf_count_to_100ns_units_ =
+        (10000000.0 / static_cast<double>(performance_frequency.QuadPart));
+  } else {
+    LOG(ERROR) <<  "High-resolution performance counters are not supported.";
+    perf_count_to_100ns_units_ = 0.0;
+  }
+}
+
+WASAPIAudioInputStream::~WASAPIAudioInputStream() {

[scherkus (not reviewing), 2011/10/18 21:12:30]

nit: close empty methods to {}

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Done.

+}
+
+bool WASAPIAudioInputStream::Open() {
+  // Verify that we are not already opened.
+  if (opened_)
+    return false;
+
+  // Obtain a reference to the IMMDevice interface of the default capturing
+  // device with the specified role.
+  HRESULT hr = SetCaptureDevice(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 = ActivateCaptureDevice();
+  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 WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
+  DCHECK(callback);
+  DCHECK(opened_);
+
+  if (!opened_)
+    return;
+
+  if (started_)
+    return;
+
+  sink_ = callback;
+
+  // Create and start the thread that will drive the capturing by waiting for
+  // capture events.
+  capture_thread_ =
+      new base::DelegateSimpleThread(this, "wasapi_capture_thread");
+  capture_thread_->Start();
+
+  // Start streaming data between the endpoint buffer and the audio engine.
+  HRESULT hr = audio_client_->Start();
+  DLOG_IF(ERROR, FAILED(hr)) << "Failed to start input streaming.";
+
+  started_ = SUCCEEDED(hr);
+}
+
+void WASAPIAudioInputStream::Stop() {
+  if (!started_)
+    return;
+
+  // 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)) {
+    LOG(ERROR) << "Failed to stop input streaming.";
+  }
+
+  // Wait until the thread completes and perform cleanup.
+  if (capture_thread_) {
+    SetEvent(stop_capture_event_.Get());
+    capture_thread_->Join();
+    capture_thread_ = NULL;
+  }
+
+  started_ = false;
+}
+
+void WASAPIAudioInputStream::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();
+  if (sink_) {
+    sink_->OnClose(this);
+    sink_ = NULL;
+  }
+
+  // Inform the audio manager that we have been closed. This will cause our
+  // destruction.
+  manager_->ReleaseInputStream(this);
+}
+
+double WASAPIAudioInputStream::HardwareSampleRate(ERole device_role) {
+  // It is assumed that this static method is called from a COM thread, i.e.,
+  // CoInitializeEx() is not called here 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: " << hr;
+    return 0.0;
+  }
+
+  ScopedComPtr<IMMDevice> endpoint_device;
+  hr = enumerator->GetDefaultAudioEndpoint(eCapture,
+                                           device_role,
+                                           endpoint_device.Receive());
+  if (FAILED(hr)) {
+    NOTREACHED() << "error code: " << 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: " << hr;
+    return 0.0;
+  }
+
+  ScopedComMem<WAVEFORMATEX> audio_engine_mix_format;
+  hr = audio_client->GetMixFormat(audio_engine_mix_format.Receive());
+  if (FAILED(hr)) {
+    NOTREACHED() << "error code: " << hr;
+    return 0.0;
+  }
+
+  return static_cast<double>(audio_engine_mix_format->nSamplesPerSec);
+}
+
+void WASAPIAudioInputStream::Run() {
+  ScopedCOMInitializerMTA com_init;
+
+  // Increase the thread priority.
+  capture_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("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 << ").";
+  }
+
+  // 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.
+  size_t buffer_frame_index = 0;
+  size_t capture_buffer_size = std::max(
+      2 * endpoint_buffer_size_frames_ * frame_size_,
+      2 * packet_size_frames_ * frame_size_);
+  scoped_array<uint8> capture_buffer(new uint8[capture_buffer_size]);
+
+  LARGE_INTEGER now_count;
+  bool recording = true;
+  HANDLE wait_array[2] = {stop_capture_event_, audio_samples_ready_event_};
+
+  while (recording) {
+    HRESULT hr = S_FALSE;
+
+    // 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_FAILED:
+        recording = false;
+        LOG(ERROR) << "WASAPI capturing failed with error code "
+                   << GetLastError();
+        break;
+      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.
+        BYTE* data_ptr = NULL;
+        UINT32 num_frames_to_read = 0;
+        DWORD flags = 0;
+        UINT64 device_position = 0;
+        UINT64 first_audio_frame_timestamp = 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,
+                                              &first_audio_frame_timestamp);
+        if (SUCCEEDED(hr)) {
+          if (num_frames_to_read != 0) {
+            size_t pos = buffer_frame_index * frame_size_;
+            size_t num_bytes = num_frames_to_read * frame_size_;
+            DCHECK_GE(capture_buffer_size, pos + num_bytes);
+
+            if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
+              // Clear out the local buffer since silence is reported.
+              memset(&capture_buffer[pos], 0, num_bytes);
+            } else {
+              // Copy captured data from audio engine buffer to local buffer.
+              memcpy(&capture_buffer[pos], data_ptr, num_bytes);
+            }
+
+            buffer_frame_index += num_frames_to_read;
+          }
+
+          hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
+          if (FAILED(hr))
+            HandleError(hr);

[scherkus (not reviewing), 2011/10/18 21:12:30]

is it safe to execute the OnError callback inside this thread?

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Good question. It actually feels like overkill to call HandleError() here. It
does not even trigger if I remove all recording devices while recording is
active.

I will modify to DLOG(ERROR) instead. Hope that is OK.

+
+          // Derive a delay estimate for the captured audio packet.
+          // The value contains two parts (A+B), where A is the delay of the
+          // first audio frame in the packet and B is the extra delay contained
+          // in any stored data. Unit is in audio frames.
+          QueryPerformanceCounter(&now_count);
+          double audio_delay_frames =
+              ((perf_count_to_100ns_units_ * now_count.QuadPart -
+                first_audio_frame_timestamp) / 10000.0) * ms_to_frame_count_ +
+                buffer_frame_index - num_frames_to_read;
+
+          // Deliver captured data to the registered consumer using a packet
+          // size which was specified at construction.
+          uint32 delay_frames = static_cast<uint32> (audio_delay_frames + 0.5);
+          while (buffer_frame_index >= packet_size_frames_) {
+            uint8* audio_data = reinterpret_cast<uint8*>(capture_buffer.get());
+
+            // Deliver data packet and delay estimation to the user.
+            sink_->OnData(this,
+                          audio_data,
+                          packet_size_bytes_,
+                          delay_frames * frame_size_);
+
+            // Store parts of the recorded data which can't be delivered
+            // using the current packet size. The stored section will be used
+            // either in the next while-loop iteration or in the next
+            // capture event.
+            memmove(&capture_buffer[0],
+                    &capture_buffer[packet_size_bytes_],
+                    (buffer_frame_index - packet_size_frames_) * frame_size_);
+
+            buffer_frame_index -= packet_size_frames_;
+            delay_frames -= packet_size_frames_;
+          }
+        }
+        break;
+    }

[scherkus (not reviewing), 2011/10/18 21:12:30]

sanity check: is there a default: condition or anything else we should be
checking for?

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

I should have covered all return cases for WaitForMultipleObjects() but adding a
default just in case.
Also modified the while() condition to make it possible to check for an invalid
exit state.

+  }
+
+  // Disable MMCSS.
+  if (mm_task && !avrt_.AvRevertMmThreadCharacteristics(mm_task)) {
+    DWORD err = GetLastError();
+    LOG(WARNING) << "Failed to disable MMCSS (error code=" << err << ").";
+  }
+}
+
+void WASAPIAudioInputStream::HandleError(HRESULT err) {
+  _com_error com_error(err);
+  std::string message(WideToUTF8(com_error.ErrorMessage()));
+  DLOG(ERROR) << "Error code: " << err;
+  NOTREACHED() << "Error details: " << message;
+
+  if (sink_)
+    sink_->OnError(this, static_cast<int>(err));
+}
+
+HRESULT WASAPIAudioInputStream::SetCaptureDevice(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 capture 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(eCapture,
+                                             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 capture device is not active.";
+        hr = E_ACCESSDENIED;
+      }
+    }
+  }
+
+  return hr;
+}
+
+HRESULT WASAPIAudioInputStream::ActivateCaptureDevice() {
+  // Creates and activates an IAudioClient COM object given the selected
+  // capture endpoint device.
+  HRESULT hr = endpoint_device_->Activate(__uuidof(IAudioClient),
+                                          CLSCTX_INPROC_SERVER,
+                                          NULL,
+                                          audio_client_.ReceiveVoid());
+  return hr;
+}
+
+HRESULT WASAPIAudioInputStream::GetAudioEngineStreamFormat() {
+  // Retrieve the stream format that the audio engine uses for its internal
+  // processing/mixing of shared-mode streams.
+  HRESULT hr = audio_client_->GetMixFormat(audio_engine_mix_format_.Receive());
+#ifndef NDEBUG

[scherkus (not reviewing), 2011/10/18 21:12:30]

ditto

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Removed.

+  if (SUCCEEDED(hr))
+    DLogFormat("Audio Engine's format:", audio_engine_mix_format_.get());
+#endif
+  return hr;
+}
+
+bool WASAPIAudioInputStream::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.
+  ScopedComMem<WAVEFORMATEX> closest_match;
+  HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
+                                                &format_,
+                                                closest_match.Receive());
+  if (hr == S_FALSE) {
+    DLOG(ERROR) << "Format is not supported but a closest match exists.";
+#ifndef NDEBUG

[scherkus (not reviewing), 2011/10/18 21:12:30]

ditto

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

Removed.

+    DLogFormat("Closest suggested capture format:", closest_match.get());
+#endif
+  }
+  return (hr == S_OK);
+}
+
+HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
+  // 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, |hnsBufferDuration| is set of 0, which ensures that the
+  // buffer is never smaller than the minimum buffer size needed to ensure
+  // that glitches do not occur between the periodic processing passes.
+  // This setting should lead to lowest possible latency.
+  HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
+                                         AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                         AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                         0,  // hnsBufferDuration
+                                         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 determines the maximum amount
+  // of capture data that the audio engine can read from 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;
+  DLOG(INFO) << "endpoint buffer size: " << endpoint_buffer_size_frames_
+             << " [frames]";
+
+#ifndef NDEBUG

[scherkus (not reviewing), 2011/10/18 21:12:30]

does this really need to be run in debug mode every time?

does it make sense to run in release mode at all (i.e., caching the values)?

[henrika (OOO until Aug 14), 2011/10/19 15:42:43]

These are trivial functions which adds value when I develop. They will come in
handy during the development of the render side as well. I would like to keep
them as is for now. Added a TODO() to be removed when all parts are in place.

I prefer getting the values each time (only called during init) since I must
create members under #ifndef NDEBUG otherwise.

[scherkus (not reviewing), 2011/10/19 17:15:55]

SGTM

+  // 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)) {
+    DLOG(INFO) << "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)) {
+    DLOG(INFO) << "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_ready_event_.Get());
+  if (FAILED(hr))
+    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(__uuidof(IAudioCaptureClient),
+                                 audio_capture_client_.ReceiveVoid());
+  return hr;
+}
Property changes on: media\audio\win\audio_low_latency_input_win.cc
___________________________________________________________________
Added: svn:eol-style
   + LF