Adding experimental exclusive-mode streaming to WASAPIAudioOutputStream

media/audio/win/audio_low_latency_output_win.cc

 // Copyright (c) 2012 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 <Functiondiscoverykeys_devpkey.h>
 
+#include "base/command_line.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"
+#include "media/base/media_switches.h"
 
 using base::win::ScopedComPtr;
 using base::win::ScopedCOMInitializer;
 
 namespace media {
 
+AUDCLNT_SHAREMODE GetShareModeImpl() {
+  const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
+  if (cmd_line->HasSwitch(switches::kEnableExclusiveMode))
+    return AUDCLNT_SHAREMODE_EXCLUSIVE;
+  else

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

nit: can remove the else

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Done.

+    return AUDCLNT_SHAREMODE_SHARED;
+}
+
+// static
+AUDCLNT_SHAREMODE WASAPIAudioOutputStream::GetShareMode() {
+  static const AUDCLNT_SHAREMODE kShareMode = GetShareModeImpl();

[no longer working on chromium, 2012/07/25 12:32:57]

do we need const for an enum?

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Why not ;-)

+  return kShareMode;
+}
+
 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),
       restart_rendering_mode_(false),
       volume_(1.0),
       endpoint_buffer_size_frames_(0),
       device_role_(device_role),
+      share_mode_(GetShareMode()),
       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";
 
+  if (AUDCLNT_SHAREMODE_EXCLUSIVE == share_mode()) {
+    DVLOG(1) << ">> Note that EXCLUSIVE MODE is enabled <<";

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

maybe VLOG?

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

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.
@@ skipping 27 matching lines @@
 WASAPIAudioOutputStream::~WASAPIAudioOutputStream() {}
 
 bool WASAPIAudioOutputStream::Open() {
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
   if (opened_)
     return true;
 
   // Create an IMMDeviceEnumerator interface and obtain a reference to
   // the IMMDevice interface of the default rendering device with the
   // specified role.
-  HRESULT hr = SetRenderDevice(device_role_);
+  HRESULT hr = SetRenderDevice();
   if (FAILED(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)) {
     return false;
   }
 
   // Retrieve the stream format which the audio engine uses for its internal
-  // processing/mixing of shared-mode streams.
+  // processing/mixing of shared-mode streams. The result of this method is
+  // ignored for shared mode streams.
   hr = GetAudioEngineStreamFormat();
   if (FAILED(hr)) {
     return false;
   }
 
   // Verify that the selected audio endpoint supports the specified format
   // set during construction.
+  // In exclusive mode, the client can choose to open the stream in any audio
+  // format that the endpoint device supports. In shared mode, the client must
+  // open the stream in the mix format that is currently in use by the audio
+  // engine (or a format that is similar to the mix format). The audio engine's
+  // input streams and the output mix from the engine are all in this format.
   if (!DesiredFormatIsSupported()) {
     return false;
   }
 
   // Initialize the audio stream between the client and the device using
-  // shared mode and a lowest possible glitch-free latency.
+  // shared or exclusive mode and a lowest possible glitch-free latency.
+  // We will enter different code paths depending on the specified share mode.
   hr = InitializeAudioEngine();
   if (FAILED(hr)) {
     return false;
   }
 
   // Register this client as an IMMNotificationClient implementation.
   // Only OnDefaultDeviceChanged() and OnDeviceStateChanged() and are
   // non-trivial.
   hr = device_enumerator_->RegisterEndpointNotificationCallback(this);
 
@@ skipping 95 matching lines @@
   // Flush all pending data and reset the audio clock stream position to 0.
   hr = audio_client_->Reset();
   if (FAILED(hr)) {
     DLOG_IF(ERROR, hr != AUDCLNT_E_NOT_INITIALIZED)
         << "Failed to reset streaming: " << std::hex << hr;
   }
 
   // Extra safety check to ensure that the buffers are cleared.
   // If the buffers are not cleared correctly, the next call to Start()
   // would fail with AUDCLNT_E_BUFFER_ERROR at IAudioRenderClient::GetBuffer().
-  UINT32 num_queued_frames = 0;
-  audio_client_->GetCurrentPadding(&num_queued_frames);
-  DCHECK_EQ(0u, num_queued_frames);
+  // This check is is only needed for shared-mode streams.
+  if (share_mode() == AUDCLNT_SHAREMODE_SHARED) {
+    UINT32 num_queued_frames = 0;
+    audio_client_->GetCurrentPadding(&num_queued_frames);
+    DCHECK_EQ(0u, num_queued_frames);
+  }
 
   // Ensure that we don't quit the main thread loop immediately next
   // time Start() is called.
   ResetEvent(stop_render_event_.Get());
 
   started_ = false;
 }
 
 void WASAPIAudioOutputStream::Close() {
   DCHECK_EQ(GetCurrentThreadId(), creating_thread_id_);
@@ skipping 26 matching lines @@
 
 void WASAPIAudioOutputStream::GetVolume(double* volume) {
   DVLOG(1) << "GetVolume()";
   *volume = static_cast<double>(volume_);
 }
 
 // static
 int 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.
+  // Note that, calling this function only makes sense for shared mode streams,
+  // since if the device will be opened in exclusive mode, then the application
+  // specified format is used instead.
   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;
   }
@@ skipping 13 matching lines @@
   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;
   }
 
+  // Retrieve the stream format that the audio engine uses for its internal
+  // processing of shared-mode streams.
   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<int>(audio_engine_mix_format->nSamplesPerSec);
 }
 
@@ skipping 14 matching lines @@
     // 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[] = { stop_render_event_,
-                          stream_switch_event_,
-                          audio_samples_render_event_ };
+  HANDLE wait_array[] = {stop_render_event_,

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

nit: revert this change? (there's still space for the closing brace and using a
space for the opening brace is pretty common)

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Fixed.

+                         stream_switch_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
@@ skipping 26 matching lines @@
           playing = false;
           error = true;
         }
         break;
       case WAIT_OBJECT_0 + 2:
         {
           // |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
+          // 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.
-          size_t num_available_frames =
-              endpoint_buffer_size_frames_ - num_queued_frames;
+          size_t num_available_frames = 0;
+
+          if (share_mode() == AUDCLNT_SHAREMODE_SHARED) {
+            // 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);
+            num_available_frames =
+                endpoint_buffer_size_frames_ - num_queued_frames;
+          } else {
+            // While the stream is running, the system alternately sends one
+            // buffer or the other to the client. This form of double buffering
+            // is referred to as "ping-ponging". Each time the client receives
+            // a buffer from the system (triggers this event) the client must
+            // process the entire buffer. Calls to the GetCurrentPadding method
+            // are unnecessary because the packet size must always equal the
+            // buffer size. In contrast to the shared mode buffering scheme,
+            // the latency for an event-driven, exclusive-mode stream depends
+            // directly on the buffer size.
+            num_available_frames = endpoint_buffer_size_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.
+          // |num_packets| will always be one for exclusive-mode streams.
           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: "
@@ skipping 80 matching lines @@
     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) {
+HRESULT WASAPIAudioOutputStream::SetRenderDevice() {
   // Create the IMMDeviceEnumerator interface.
   HRESULT hr =  CoCreateInstance(__uuidof(MMDeviceEnumerator),
                                  NULL,
                                  CLSCTX_INPROC_SERVER,
                                  __uuidof(IMMDeviceEnumerator),
                                  device_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 = device_enumerator_->GetDefaultAudioEndpoint(
-        eRender, device_role, endpoint_device_.Receive());
+        eRender, device_role_, endpoint_device_.Receive());
     if (FAILED(hr))
       return hr;
 
     // 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.";
@@ skipping 15 matching lines @@
   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() {
+  // Determine, before calling IAudioClient::Initialize, whether the audio

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

nit: ()

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Done.

+  // engine supports a particular stream format.
   // 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.
+  // which is stored in the |audio_engine_mix_format_| member.
   base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
-  HRESULT hr = audio_client_->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
+  HRESULT hr = audio_client_->IsFormatSupported(share_mode(),
                                                 &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: "
+    DVLOG(1) << "shared mode (default) device period: "
              << static_cast<double>(default_device_period / 10000.0)
              << " [ms]";
     // Exclusive mode device period.
-    DVLOG(1) << "minimum device period: "
+    DVLOG(1) << "exclusive mode (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
 
+  HRESULT hr = S_FALSE;
+  REFERENCE_TIME requested_buffer_duration = 0;
+
+  // Perform different initialization depending on if the device shall be
+  // opened in shared mode or in exclusive mode.
+  if (share_mode() == AUDCLNT_SHAREMODE_SHARED) {
+    // The device will be opened in shared mode and use the WAS format.

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

what about pulling these two initialization paths out into separate functions to
make this one more easily readable?

+
+    // 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;
+    requested_buffer_duration =
+        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.
+    hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
+                                   AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                   AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                   requested_buffer_duration,
+                                   0,
+                                   &format_,
+                                   NULL);
+  } else {
+    // The device will be opened in exclusive mode and use the application
+    // specified format.
+
+    float f = (1000.0 * packet_size_frames_) / format_.nSamplesPerSec;
+    requested_buffer_duration = static_cast<REFERENCE_TIME>(f*10000.0 + 0.5);
+
+    // Initialize the audio stream between the client and the device.
+    // For an exclusive-mode stream that uses event-driven buffering, the
+    // caller must specify nonzero values for hnsPeriodicity and
+    // hnsBufferDuration, and the values of these two parameters must be equal.
+    // The Initialize method allocates two buffers for the stream. Each buffer
+    // is equal in duration to the value of the hnsBufferDuration parameter.
+    // Following the Initialize call for a rendering stream, the caller should
+    // fill the first of the two buffers before starting the stream.
+    hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE,
+                                   AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                   AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                   requested_buffer_duration,
+                                   requested_buffer_duration,
+                                   &format_,
+                                   NULL);
+    if (FAILED(hr)) {
+      if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
+        DLOG(ERROR) << "AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED";
+
+        UINT32 aligned_buffer_size = 0;
+        audio_client_->GetBufferSize(&aligned_buffer_size);
+        DVLOG(1) << "Use aligned buffer size instead: " << aligned_buffer_size;
+        audio_client_.Release();
+
+        // Calculate new aligned periodicity. Each unit of reference time
+        // is 100 nanoseconds.
+        REFERENCE_TIME aligned_buffer_duration = static_cast<REFERENCE_TIME>(
+            10000000.0 * aligned_buffer_size / format_.nSamplesPerSec + 0.5);
+
+        // It is possible to re-activate and re-initialize the audio client
+        // at this stage but we bail out with an error code instead and
+        // combine it with a log message which informs about the suggested
+        // aligned buffer size which should be used instead.
+        DVLOG(1) << "aligned_buffer_duration: "
+                 << static_cast<double>(aligned_buffer_duration / 10000.0)
+                 << " [ms]";
+      } else if (hr == AUDCLNT_E_INVALID_DEVICE_PERIOD) {
+        // We will get this error if we try to use a smaller buffer size than
+        // the minimum supported size (usually ~3ms on Windows 7).
+        DLOG(ERROR) << "AUDCLNT_E_INVALID_DEVICE_PERIOD";
+      }
+    }
+  }
+
+  if (FAILED(hr)) {
+    DVLOG(1) << "IAudioClient::Initialize() failed: " << std::hex << hr;

[tommi (sloooow) - chröme, 2012/07/25 11:49:40]

should this perhaps be LOG(WARNING)?

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Changed to PLOG(WARNING)

+    return hr;
+  }
+
+  // Retrieve the length of the endpoint buffer. The buffer length represents
+  // 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]";
+
+  // The buffer scheme for exclusive mode streams is not designed for max
+  // flexibility. We only allow a "perfect match" between the packet size set
+  // by the user and the actual endpoint buffer size.
+  if (share_mode() == AUDCLNT_SHAREMODE_EXCLUSIVE) {

[no longer working on chromium, 2012/07/25 12:32:57]

combine two if:
if (share_mode() == AUDCLNT_SHAREMODE_EXCLUSIVE &&
    endpoint_buffer_size_frames_ != packet_size_frames_)

[henrika (OOO until Aug 14), 2012/07/25 15:26:30]

Done.

+    if (endpoint_buffer_size_frames_ != packet_size_frames_) {
+      hr = AUDCLNT_E_INVALID_SIZE;
+      DLOG(ERROR) << "AUDCLNT_E_INVALID_SIZE";
+      return hr;
+    }
+  }
+
   // 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.
@@ skipping 160 matching lines @@
   // are now re-initiated and it is now possible to re-start audio rendering.
 
   // Start rendering again using the new default audio endpoint.
   hr = audio_client_->Start();
 
   restart_rendering_mode_ = false;
   return SUCCEEDED(hr);
 }
 
 }  // namespace media