Adds support for multi-channel output audio for the low-latency path in Windows.

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;
 using base::win::ScopedCoMem;
 
 namespace media {
 
+typedef uint32 ChannelConfig;
+
+// Ensure that the alignment of members will be on a boundary that is a
+// multiple of 1 byte.
+#pragma pack(push)
+#pragma pack(1)
+
+struct LayoutMono_16bit {
+  int16 center;
+};
+
+struct LayoutStereo_16bit {
+  int16 left;
+  int16 right;
+};
+
+struct Layout7_1_16bit {
+  int16 front_left;
+  int16 front_right;
+  int16 front_center;
+  int16 low_frequency;
+  int16 back_left;
+  int16 back_right;
+  int16 side_left;
+  int16 side_right;
+};
+
+#pragma pack(pop)
+
+// Retrieves the stream format that the audio engine uses for its internal
+// processing/mixing of shared-mode streams.
+static HRESULT GetMixFormat(ERole device_role, WAVEFORMATEX** device_format) {
+  // Note that we are using the IAudioClient::GetMixFormat() API to get the
+  // device format in this function. It is in fact possible to be "more native",
+  // and ask the endpoint device directly for its properties. Given a reference
+  // to the IMMDevice interface of an endpoint object, a client can obtain a
+  // reference to the endpoint object's property store by calling the
+  // IMMDevice::OpenPropertyStore() method. However, I have not been able to
+  // access any valuable information using this method on my HP Z600 desktop,
+  // hence it feels more appropriate to use the IAudioClient::GetMixFormat()
+  // approach instead.
+
+  // 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. However, the result of this method can
+  // be useful for testing purposes so we don't DCHECK here.
+  DLOG_IF(WARNING, WASAPIAudioOutputStream::GetShareMode() ==
+          AUDCLNT_SHAREMODE_EXCLUSIVE) <<
+      "The mixing sample rate will be ignored for exclusive-mode streams.";
+
+  // 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),

[scherkus (not reviewing), 2012/08/02 17:14:14]

remove extra space between = and CCI()

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+                                 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());
+  DCHECK(SUCCEEDED(hr)) << "Failed to activate device: " << std::hex << hr;
+  if (SUCCEEDED(hr)) {
+    hr = audio_client->GetMixFormat(device_format);
+    DCHECK(SUCCEEDED(hr)) << "GetMixFormat: " << std::hex << hr;
+  }
+
+  return hr;
+}
+
+// Retrieves an integer mask which corresponds to the channel layout the
+// audio engine uses for its internal processing/mixing of shared-mode
+// streams. This mask indicates which channels are present in the multi-
+// channel stream. The least significant bit corresponds with the Front Left
+// speaker, the next least significant bit corresponds to the Front Right
+// speaker, and so on, continuing in the order defined in KsMedia.h.
+// See http://msdn.microsoft.com/en-us/library/windows/hardware/ff537083(v=vs.85).aspx
+// for more details.
+static ChannelConfig GetChannelConfig() {
+  // Use a WAVEFORMATEXTENSIBLE structure since it can specify both the
+  // number of channels and the mapping of channels to speakers for
+  // multichannel devices.
+  base::win::ScopedCoMem<WAVEFORMATPCMEX> format_ex;
+  HRESULT hr = GetMixFormat(
+    eConsole, reinterpret_cast<WAVEFORMATEX**>(&format_ex));

[scherkus (not reviewing), 2012/08/02 17:14:14]

indent 2 more spaces

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+  if (FAILED(hr))
+    return 0;
+
+  // The dwChannelMask member specifies which channels are present in the
+  // multichannel stream. The least significant bit corresponds to the
+  // front left speaker, the next least significant bit corresponds to the
+  // front right speaker, and so on.
+  // See http://msdn.microsoft.com/en-us/library/windows/desktop/dd757714(v=vs.85).aspx
+  // for more details on the channel mapping.
+  DVLOG(2) << "dwChannelMask: 0x" << std::hex << format_ex->dwChannelMask;
+
+#if !defined(NDEBUG)
+  // See http://en.wikipedia.org/wiki/Surround_sound for more details on
+  // how to name various speaker configurations. The list below is not complete.
+  const char* speaker_config("Undefined");

[scherkus (not reviewing), 2012/08/02 17:14:14]

() notation is a bit odd ... use = instead

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+  switch (format_ex->dwChannelMask) {
+    case KSAUDIO_SPEAKER_MONO:
+      speaker_config = "Mono";
+      break;
+    case KSAUDIO_SPEAKER_STEREO:
+      speaker_config = "Stereo";
+      break;
+    case KSAUDIO_SPEAKER_5POINT1_SURROUND:
+      speaker_config = "5.1 surround";
+      break;
+    case  KSAUDIO_SPEAKER_5POINT1:

[scherkus (not reviewing), 2012/08/02 17:14:14]

remove extra spaces

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+      speaker_config = "5.1";
+      break;
+    case  KSAUDIO_SPEAKER_7POINT1_SURROUND:

[scherkus (not reviewing), 2012/08/02 17:14:14]

remove extra spaces

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+      speaker_config = "7.1 surround";
+      break;
+    case KSAUDIO_SPEAKER_7POINT1:
+      speaker_config = "7.1";
+      break;
+    default:
+      break;
+  }
+  DVLOG(2) << "speaker configuration: " << speaker_config;
+#endif
+
+  return static_cast<ChannelConfig>(format_ex->dwChannelMask);
+}
+
+// Converts Microsoft's channel configuration to ChannelLayout.
+// This mapping is not perfect but the best we can do given the current
+// ChannelLayout enumerator and the Windows-specific speaker configurations
+// defined in ksmedia.h. Don't assume that the channel ordering in
+// ChannelLayout is exactly the same as the Windows specific configuration.
+// As an example: KSAUDIO_SPEAKER_7POINT1_SURROUND is mapped to
+// CHANNEL_LAYOUT_7_1 but the positions of Back L, Back R and Side L, Side R
+// speakers are different in these two definitions.
+static ChannelLayout ChannelConfigToChannelLayout(ChannelConfig config) {
+  switch (config) {
+    case KSAUDIO_SPEAKER_DIRECTOUT:
+      return CHANNEL_LAYOUT_NONE;
+    case KSAUDIO_SPEAKER_MONO:
+      return CHANNEL_LAYOUT_MONO;
+    case KSAUDIO_SPEAKER_STEREO:
+      return CHANNEL_LAYOUT_STEREO;
+    case KSAUDIO_SPEAKER_QUAD:
+      return CHANNEL_LAYOUT_QUAD;
+    case KSAUDIO_SPEAKER_SURROUND:
+      return CHANNEL_LAYOUT_4_0;
+    case KSAUDIO_SPEAKER_5POINT1:
+      return CHANNEL_LAYOUT_5_1_BACK;
+    case KSAUDIO_SPEAKER_5POINT1_SURROUND:
+      return CHANNEL_LAYOUT_5_1;
+    case KSAUDIO_SPEAKER_7POINT1:
+      return CHANNEL_LAYOUT_7_1_WIDE;
+    case KSAUDIO_SPEAKER_7POINT1_SURROUND:
+      return CHANNEL_LAYOUT_7_1;
+    default:
+      DVLOG(1) << "Unsupported channel layout: " << config;
+      return CHANNEL_LAYOUT_UNSUPPORTED;
+  }
+}
+
+// mono/stereo -> N.1 up-mixing where N=out_channels-1.
+// See http://www.w3.org/TR/webaudio/#UpMix-sub for details.
+// TODO(henrika): use ChannelLayout for channel parameters.
+// TODO(henrika): can we do this in-place by processing the samples in
+// reverse order when sizeof(out) > sizeof(in) (upmixing)?
+static int ChannelUpMix(void* input,

[scherkus (not reviewing), 2012/08/02 17:14:14]

man this function is going to get massive over time -- any ideas on how to
parameterize things?

also won't this break if we do 8-bit or 24-bit audio?

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

I have ideas on how to make it smaller, yes. Have no time to finalize today
before I go on vacation. Any input from your side is of course appreciated.

Regarding 8-bit and 24-bit audio:

The old implementation has not been tested at all for 8-bits but there is
nothing in there that prevents it from supporting 8 bits. To my knowledge, no
low-latency client requires 8-bit support today and I have not written any unit
tests for it. 24-bits will not work since it lacks fundamental parts to support
it. One example: AdjustVolume() does not support 24-bit.

As you point out, the new implementation will not work for 8-bits if up-mixing
is required. Same for 24-bit.

Best I can do for now is to add a error logging for it.

[henrika (OOO until Aug 14), 2012/08/03 15:18:34]

More on the 8-bit case. I am actually not sure that we should ever open in 8-bit
mode since I don't think that any device supports is as default mode (i.e., user
can't set it in his device properties). Hence, the more correct way should be to
open up at 16 and (same as we could report back to the client). Then if the
client feeds 8-bit we should do 8->16 bit conversion, or better, the client does
this and feeds converted 16-bit samples.

24-bit is another story since the user can sometimes select that and it will
then be the preferred bit depth.

+                        void* output,
+                        int in_channels,
+                        int out_channels,
+                        size_t number_of_input_bytes) {
+  DCHECK_GT(out_channels, in_channels);
+
+  const int kChannelRatio = out_channels / in_channels;
+
+  // 1 -> 2
+  if (in_channels == 1 && out_channels == 2) {
+    LayoutMono_16bit* in = reinterpret_cast<LayoutMono_16bit*>(input);
+    LayoutStereo_16bit* out = reinterpret_cast<LayoutStereo_16bit*>(output);
+    int number_of_input_mono_samples = (number_of_input_bytes >> 1);
+
+    // Copy same input mono sample to both output channels.
+    for (int i = 0; i < number_of_input_mono_samples; ++i) {
+      out->left = in->center;
+      out->right = in->center;
+      in++;
+      out++;
+    }
+
+    return (kChannelRatio * number_of_input_bytes);
+  }
+
+  // 1 -> 7.1
+  if (in_channels == 1 && out_channels == 8) {
+    LayoutMono_16bit* in = reinterpret_cast<LayoutMono_16bit*>(input);
+    Layout7_1_16bit* out = reinterpret_cast<Layout7_1_16bit*>(output);
+    int number_of_input_mono_samples = (number_of_input_bytes >> 1);
+
+    // Zero out all frames first.
+    memset(out, 0, number_of_input_mono_samples * sizeof(out[0]));
+
+    // Copy input sample to output center channel.
+    for (int i = 0; i < number_of_input_mono_samples; ++i) {
+      out->front_center = in->center;
+      in++;
+      out++;
+    }
+
+    return (kChannelRatio * number_of_input_bytes);
+  }
+
+  // 2 -> 7.1
+  if (in_channels == 2 && out_channels == 8) {
+    LayoutStereo_16bit* in = reinterpret_cast<LayoutStereo_16bit*>(input);
+    Layout7_1_16bit* out = reinterpret_cast<Layout7_1_16bit*>(output);
+    int number_of_input_stereo_samples = (number_of_input_bytes >> 2);
+
+    // Zero out all frames first.
+    memset(out, 0, number_of_input_stereo_samples * sizeof(out[0]));
+
+    // Copy left and right input channels to the same output channels.
+    for (int i = 0; i < number_of_input_stereo_samples; ++i) {
+      out->front_left = in->left;
+      out->front_right = in->right;
+      in++;
+      out++;
+    }
+
+    return (kChannelRatio * number_of_input_bytes);
+  }
+
+  LOG(ERROR) << "Up-mixing " << in_channels << "->"
+             << out_channels << " is not supported.";
+  return 0;
+}
+
 // static
 AUDCLNT_SHAREMODE WASAPIAudioOutputStream::GetShareMode() {
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
   if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio))
     return AUDCLNT_SHAREMODE_EXCLUSIVE;
   return AUDCLNT_SHAREMODE_SHARED;
 }
 
 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()),
+      client_channel_count_(params.channels()),
       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 (share_mode() == AUDCLNT_SHAREMODE_EXCLUSIVE) {
+  if (share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE) {
     VLOG(1) << ">> Note that EXCLUSIVE MODE is enabled <<";
   }
 
-  // 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;
+  // Set up the desired render format specified by the client. We use the
+  // WAVE_FORMAT_EXTENSIBLE structure to ensure that multiple channel ordering
+  // and high precision data can be supported.
+
+  // Begin with the WAVEFORMATEX structure that specifies the basic format.
+  WAVEFORMATEX* format = &format_.Format;
+  format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+  format->nChannels =  HardwareChannelCount();
+  format->nSamplesPerSec = params.sample_rate();
+  format->wBitsPerSample = params.bits_per_sample();
+  format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels;
+  format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign;
+  format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+
+  // Add the parts which are unique to WAVE_FORMAT_EXTENSIBLE.
+  format_.Samples.wValidBitsPerSample = params.bits_per_sample();
+  format_.dwChannelMask = GetChannelConfig();
+  format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
 
   // Size in bytes of each audio frame.
-  frame_size_ = format_.nBlockAlign;
+  frame_size_ = format->nBlockAlign;
+
+  // It is possible to set the number of channels in |params| to a lower value
+  // than we use as the internal number of audio channels when the audio stream
+  // is opened. If this mode (channel_factor_ > 1) is set, the native audio
+  // layer will expect a larger number of channels in the interleaved audio
+  // stream and a channel up-mix will be performed after the OnMoreData()
+  // callback to compensate for the lower number of channels provided by the
+  // audio source.
+  // Example: params.channels() is 2 and endpoint_channel_count() is 8 =>
+  // the audio stream is opened up in 7.1 surround mode but the source only
+  // provides a stereo signal as input, i.e., a stereo up-mix (2 -> 7.1) will
+  // take place before sending the stream to the audio driver.
+  DCHECK_GE(channel_factor(), 1) << "Unsupported channel count.";
+  DVLOG(1) << "client channels (in): " << params.channels();
+  DVLOG(1) << "channel factor: " << channel_factor();
 
   // 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.GetBytesPerBuffer() / format_.nBlockAlign;
-  packet_size_bytes_ = params.GetBytesPerBuffer();
+  packet_size_frames_ =
+      (channel_factor() * params.GetBytesPerBuffer()) / format->nBlockAlign;
+  packet_size_bytes_ = channel_factor() * params.GetBytesPerBuffer();
   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));
@@ skipping 153 matching lines @@
   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().
   // This check is is only needed for shared-mode streams.
-  if (share_mode() == AUDCLNT_SHAREMODE_SHARED) {
+  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;
@@ skipping 27 matching lines @@
   }
   volume_ = volume_float;
 }
 
 void WASAPIAudioOutputStream::GetVolume(double* volume) {
   DVLOG(1) << "GetVolume()";
   *volume = static_cast<double>(volume_);
 }
 
 // static
+int WASAPIAudioOutputStream::HardwareChannelCount() {
+  // Use a WAVEFORMATEXTENSIBLE structure since it can specify both the
+  // number of channels and the mapping of channels to speakers for
+  // multichannel devices.
+  base::win::ScopedCoMem<WAVEFORMATPCMEX> format_ex;
+  HRESULT hr = GetMixFormat(
+      eConsole, reinterpret_cast<WAVEFORMATEX**>(&format_ex));
+  if (FAILED(hr))
+    return 0;
+
+  // Number of channels in the stream. Corresponds to the number of bits
+  // set in the dwChannelMask.
+  DVLOG(1) << "endpoint channels (out): " << format_ex->Format.nChannels;
+
+  return static_cast<int>(format_ex->Format.nChannels);
+}
+
+// static
+ChannelLayout WASAPIAudioOutputStream::HardwareChannelLayout() {
+  return ChannelConfigToChannelLayout(GetChannelConfig());
+}
+
+// static
 int WASAPIAudioOutputStream::HardwareSampleRate(ERole device_role) {
-  // 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. However, the result of this method can
-  // be useful for testing purposes so we don't DCHECK here.
-  DLOG_IF(WARNING, GetShareMode() == AUDCLNT_SHAREMODE_EXCLUSIVE) <<
-      "The mixing sample rate will be ignored for exclusive-mode streams.";
+  base::win::ScopedCoMem<WAVEFORMATEX> format;
+  HRESULT hr = GetMixFormat(device_role, &format);
+  if (FAILED(hr))
+    return 0;
 
-  // 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;
-  }
-
-  // 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);
+  DVLOG(2) << "nSamplesPerSec: " << format->nSamplesPerSec;
+  return static_cast<int>(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.
@@ skipping 60 matching lines @@
         {
           // |audio_samples_render_event_| has been set.
           UINT32 num_queued_frames = 0;
           uint8* audio_data = NULL;
 
           // 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 = 0;
 
-          if (share_mode() == AUDCLNT_SHAREMODE_SHARED) {
+          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
@@ skipping 31 matching lines @@
             // 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 *
+              double pos_sample_playing_frames = format_.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(
-                audio_data, packet_size_bytes_,
-                AudioBuffersState(0, audio_delay_bytes));
+
+            uint32 num_filled_bytes = 0;
+
+            if (channel_factor() == 1) {
+              // Case I: no up-mixing.
+              num_filled_bytes = source_->OnMoreData(
+                  audio_data, packet_size_bytes_,
+                  AudioBuffersState(0, audio_delay_bytes));
+            } else {
+              // Case II: up-mixing.
+              const int audio_source_size_bytes =
+                  packet_size_bytes_ / channel_factor();
+              scoped_array<uint8> buffer;
+              buffer.reset(new uint8[audio_source_size_bytes]);
+
+              num_filled_bytes = source_->OnMoreData(
+                  buffer.get(), audio_source_size_bytes,
+                  AudioBuffersState(0, audio_delay_bytes));
+
+              num_filled_bytes = ChannelUpMix(buffer.get(),
+                                              &audio_data[0],
+                                              client_channel_count_,
+                                              endpoint_channel_count(),
+                                              num_filled_bytes);
+            }
 
             // Perform in-place, software-volume adjustments.
+            // TODO(henrika): it is possible to adjust the volume in the
+            // ChannelUpMix() function.
             media::AdjustVolume(audio_data,
                                 num_filled_bytes,
-                                format_.nChannels,
-                                format_.wBitsPerSample >> 3,
+                                endpoint_channel_count(),
+                                format_.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));
+                     (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_;
           }
         }
@@ skipping 69 matching lines @@
   // 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());
   if (SUCCEEDED(hr)) {
     // Retrieve the stream format that the audio engine uses for its internal
     // processing/mixing of shared-mode streams.
     audio_engine_mix_format_.Reset(NULL);
-    hr = audio_client->GetMixFormat(&audio_engine_mix_format_);
+    hr = audio_client->GetMixFormat(
+        reinterpret_cast<WAVEFORMATEX**>(&audio_engine_mix_format_));
 
     if (SUCCEEDED(hr)) {
       audio_client_ = audio_client;
     }
   }
 
   return hr;
 }
 
 bool WASAPIAudioOutputStream::DesiredFormatIsSupported() {
   // Determine, before calling IAudioClient::Initialize(), whether the audio
   // 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 and it is also
   // possible to receive a proposed (closest) format if the current format is
   // not supported.
-  base::win::ScopedCoMem<WAVEFORMATEX> closest_match;
-  HRESULT hr = audio_client_->IsFormatSupported(share_mode(),
-                                                &format_,
-                                                &closest_match);
+  base::win::ScopedCoMem<WAVEFORMATEXTENSIBLE> closest_match;
+  HRESULT hr = audio_client_->IsFormatSupported(
+    share_mode_, reinterpret_cast<WAVEFORMATEX*>(&format_),

[scherkus (not reviewing), 2012/08/02 17:14:14]

indent 2 more spaces

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

+    reinterpret_cast<WAVEFORMATEX**>(&closest_match));
 
   // This log can only be triggered for shared mode.
   DLOG_IF(ERROR, hr == S_FALSE) << "Format is not supported "
                                 << "but a closest match exists.";
   // This log can be triggered both for shared and exclusive modes.
   DLOG_IF(ERROR, hr == AUDCLNT_E_UNSUPPORTED_FORMAT) << "Unsupported format.";
   if (hr == S_FALSE) {
-    DVLOG(1) << "wFormatTag    : " << closest_match->wFormatTag;
-    DVLOG(1) << "nChannels     : " << closest_match->nChannels;
-    DVLOG(1) << "nSamplesPerSec: " << closest_match->nSamplesPerSec;
-    DVLOG(1) << "wBitsPerSample: " << closest_match->wBitsPerSample;
+    DVLOG(1) << "wFormatTag    : " << closest_match->Format.wFormatTag;
+    DVLOG(1) << "nChannels     : " << closest_match->Format.nChannels;
+    DVLOG(1) << "nSamplesPerSec: " << closest_match->Format.nSamplesPerSec;
+    DVLOG(1) << "wBitsPerSample: " << closest_match->Format.wBitsPerSample;
   }
 
   return (hr == S_OK);
 }
 
 HRESULT WASAPIAudioOutputStream::InitializeAudioEngine() {
 #if !defined(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
@@ skipping 21 matching lines @@
       DVLOG(1) << "stream latency: " << static_cast<double>(latency / 10000.0)
                << " [ms]";
     }
   }
 #endif
 
   HRESULT hr = S_FALSE;
 
   // Perform different initialization depending on if the device shall be
   // opened in shared mode or in exclusive mode.
-  hr = (share_mode() == AUDCLNT_SHAREMODE_SHARED) ?
+  hr = (share_mode_ == AUDCLNT_SHAREMODE_SHARED) ?
       SharedModeInitialization() : ExclusiveModeInitialization();
   if (FAILED(hr)) {
     LOG(WARNING) << "IAudioClient::Initialize() failed: " << std::hex << hr;
     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 &&
+  if (share_mode_ == AUDCLNT_SHAREMODE_EXCLUSIVE &&
       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.
   hr = audio_client_->GetService(__uuidof(IAudioRenderClient),
                                  audio_render_client_.ReceiveVoid());
   return hr;
 }
 
 HRESULT WASAPIAudioOutputStream::SharedModeInitialization() {
-  DCHECK_EQ(share_mode(), AUDCLNT_SHAREMODE_SHARED);
+  DCHECK_EQ(share_mode_, AUDCLNT_SHAREMODE_SHARED);
 
   // 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) {
+  if (audio_engine_mix_format_->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) {
+  } else if (audio_engine_mix_format_->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 =
       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,
-                                         0,
-                                         &format_,
-                                         NULL);
+  HRESULT hr = S_FALSE;
+  hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_SHARED,
+                                 AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                 AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                 requested_buffer_duration,
+                                 0,
+                                 reinterpret_cast<WAVEFORMATEX*>(&format_),
+                                 NULL);
   return hr;
 }
 
 HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization() {
-  DCHECK_EQ(share_mode(), AUDCLNT_SHAREMODE_EXCLUSIVE);
+  DCHECK_EQ(share_mode_, AUDCLNT_SHAREMODE_EXCLUSIVE);
 
-  float f = (1000.0 * packet_size_frames_) / format_.nSamplesPerSec;
+  float f = (1000.0 * packet_size_frames_) / format_.Format.nSamplesPerSec;
   REFERENCE_TIME 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.
-  HRESULT hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE,
-                                         AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
-                                         AUDCLNT_STREAMFLAGS_NOPERSIST,
-                                         requested_buffer_duration,
-                                         requested_buffer_duration,
-                                         &format_,
-                                         NULL);
+  HRESULT hr = S_FALSE;
+  hr = audio_client_->Initialize(AUDCLNT_SHAREMODE_EXCLUSIVE,
+                                 AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
+                                 AUDCLNT_STREAMFLAGS_NOPERSIST,
+                                 requested_buffer_duration,
+                                 requested_buffer_duration,
+                                 reinterpret_cast<WAVEFORMATEX*>(&format_),
+                                 NULL);
   if (FAILED(hr)) {
     if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
       LOG(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);
+          (10000000.0 * aligned_buffer_size / format_.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
@@ skipping 19 matching lines @@
   NOTREACHED() << "IMMNotificationClient should not use this method.";
   if (iid == IID_IUnknown || iid == __uuidof(IMMNotificationClient)) {
     *object = static_cast < IMMNotificationClient*>(this);
   } else {
     return E_NOINTERFACE;
   }
   return S_OK;
 }
 
 STDMETHODIMP WASAPIAudioOutputStream::OnDeviceStateChanged(LPCWSTR device_id,
-                                                      DWORD new_state) {
+    DWORD new_state) {

[scherkus (not reviewing), 2012/08/02 17:14:14]

nit: this should be aligned at the (

if it doesn't fit then drop device_id to this line as well

[henrika (OOO until Aug 14), 2012/08/03 14:55:56]

Done.

 #ifndef NDEBUG
   std::string device_name = GetDeviceName(device_id);
   std::string device_state;
 
   switch (new_state) {
     case DEVICE_STATE_ACTIVE:
       device_state = "ACTIVE";
       break;
     case DEVICE_STATE_DISABLED:
       device_state = "DISABLED";
@@ skipping 123 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