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 ChannelConfigMask;
+
+// 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 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.
+HRESULT GetMixFormat(ERole device_role, WAVEFORMATEX** device_format) {

[scherkus (not reviewing), 2012/08/02 00:41:57]

static

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done. However, is it really needed when I am in the media namespace?

+  // 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),
+                                 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.
+ChannelConfigMask ChannelConfig() {

[scherkus (not reviewing), 2012/08/02 00:41:57]

static

[scherkus (not reviewing), 2012/08/02 00:41:57]

ChannelConfig() isn't a terribly good message name -- what about
GetChannelConfig() ?

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done.

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done.

+  // 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;
+
+  // 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");
+  switch (format_ex->dwChannelMask) {
+  case KSAUDIO_SPEAKER_MONO:
+    speaker_config = "Mono";
+  case KSAUDIO_SPEAKER_STEREO:
+    speaker_config = "Stereo";
+  case KSAUDIO_SPEAKER_5POINT1_SURROUND:
+    speaker_config = "5.1 surround";
+  case  KSAUDIO_SPEAKER_5POINT1:
+    speaker_config = "5.1";
+  case  KSAUDIO_SPEAKER_7POINT1_SURROUND:
+    speaker_config = "7.1 surround";
+  case KSAUDIO_SPEAKER_7POINT1:
+    speaker_config = "7.1";
+  }
+  DVLOG(2) << "speaker configuration: " << speaker_config;
+#endif
+
+  return static_cast<ChannelConfigMask>(format_ex->dwChannelMask);
+}
+
+// Converts Microsoft's channel configuration to Chrome's 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.
+ChannelLayout ChannelConfigToChromeChannelLayout(ChannelConfigMask config) {

[scherkus (not reviewing), 2012/08/02 00:41:57]

nit: drop the Chrome part from function name -- having ChannelLayout (which is a
type) is good enough

ditto for comment

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Got it. Took it from FFmpeg ;-)
Removed Mask from ChannelConfigMask typedef as well. Give cleaner code I think.

BTW - why isn't ChannelLayout in media?

+    switch (config) {

[scherkus (not reviewing), 2012/08/02 00:41:57]

fix indent

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done.

+    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;
+    }
+}
+
+// 2 -> N.1 up-mixing where N=out_channels-1.
+// See http://www.w3.org/TR/webaudio/#UpMix-sub for details.
+// TODO(henrika): improve comment and possible use ChannelLayout for channel
+// parameters.
+bool ChannelUpMix(void* input,
+                  void* output,
+                  int in_channels,
+                  int out_channels,
+                  size_t number_of_input_bytes) {
+  DCHECK(input);

[scherkus (not reviewing), 2012/08/02 00:41:57]

the DCHECKs for input/output aren't needed

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done.

+  DCHECK(output);
+  DCHECK_GT(out_channels, in_channels);
+
+  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.
+    // TODO(henrika): can we do this in-place by processing the samples in
+    // reverse order when sizeof(out) > sizeof(in) (upmixing)?
+    for (int i = 0; i < number_of_input_stereo_samples; ++i) {
+      out->front_left = in->left;
+      out->front_right = in->right;
+      in++;
+      out++;
+    }
+  } else {
+    LOG(ERROR) << "Up-mixing is not supported.";

[scherkus (not reviewing), 2012/08/02 00:41:57]

document # of channels?

[henrika (OOO until Aug 14), 2012/08/02 09:01:16]

Done.

+    return false;
+  }
+  return true;
+}
+
 // 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 = 22;
+
+  // Add the parts which are unique to WAVE_FORMAT_EXTENSIBLE.
+  format_.Samples.wValidBitsPerSample = params.bits_per_sample();
+  format_.dwChannelMask = ChannelConfig();
+  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: " << 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(2) << "endpoint channels: " << format_ex->Format.nChannels;
+
+  return static_cast<int>(format_ex->Format.nChannels);
+}
+
+// static
+ChannelLayout WASAPIAudioOutputStream::HardwareChannelLayout() {
+  return ChannelConfigToChromeChannelLayout(ChannelConfig());
+}
+
+// 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));
 
-            // Perform in-place, software-volume adjustments.
-            media::AdjustVolume(audio_data,
-                                num_filled_bytes,
-                                format_.nChannels,
-                                format_.wBitsPerSample >> 3,
-                                volume_);
+            // TODO(henrika): improve comments about possible upmixing here...
 
-            // 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));
+            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));
+
+              // Perform in-place, software-volume adjustments.
+              media::AdjustVolume(audio_data,
+                                  num_filled_bytes,
+                                  format_.Format.nChannels,
+                                  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));
+              }
+            } 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));
+
+              ChannelUpMix(buffer.get(),
+                           &audio_data[0],
+                           client_channel_count_,
+                           endpoint_channel_count(),
+                           num_filled_bytes);
+
+              // TODO(henrika): take care of zero-out for this case as well.
             }
 
             // 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_),
+    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_,
+                                         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_,
+                                         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) {
 #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