Support floating-point audio output for Windows7+

media/audio/win/audio_low_latency_output_win.cc

Index: media/audio/win/audio_low_latency_output_win.cc
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index 847a5a2b029bc08dbf0999fa852098b82053576e..f9ae3eb498102d3e074bf9ff78b2caf4e8ee6266 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -18,6 +18,7 @@
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
+#include "media/base/audio_sample_types.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 
@@ -72,7 +73,14 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
       share_mode_(GetShareMode()),
       num_written_frames_(0),
       source_(NULL),
-      audio_bus_(AudioBus::Create(params)) {
+      audio_bus_(AudioBus::Create(AudioParameters(
+          params.format(),
+          params.channel_layout(),
+          params.sample_rate(),
+          // Ignore the given bits per sample; we want 32 because
+          // we're outputting floats.
+          32,
+          params.frames_per_buffer()))) {
   DCHECK(manager_);
 
   // The empty string is used to indicate a default device and the
@@ -98,15 +106,16 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
   format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
   format->nChannels = params.channels();
   format->nSamplesPerSec = params.sample_rate();
-  format->wBitsPerSample = params.bits_per_sample();
+  // Always want 32 bits because we're using floats.
+  format->wBitsPerSample = 32;

[tommi (sloooow) - chröme, 2016/11/02 20:23:04]

nit: Add a constant at the top of the file for this? Just so that readers will
be more aware of where this value needs to be set.

[Raymond Toy, 2016/11/02 21:26:14]

Done. Please up choose a good name if you don't like kBitsPerFloatSample.

   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_.Samples.wValidBitsPerSample = 32;
   format_.dwChannelMask = CoreAudioUtil::GetChannelConfig(device_id, eRender);
-  format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+  format_.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;

[tommi (sloooow) - chröme, 2016/11/02 20:23:04]

that's it? :-|

[Raymond Toy, 2016/11/02 21:26:14]

I think so. :-)  I can hear output now that wasn't heard before on my windows
machine.

 
   // Store size (in different units) of audio packets which we expect to
   // get from the audio endpoint device in each render event.
@@ -549,8 +558,8 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     // source such as NaCl.
     const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
     audio_bus_->Scale(volume_);
-    audio_bus_->ToInterleaved(
-        frames_filled, bytes_per_sample, audio_data);
+    audio_bus_->ToInterleaved<Float32SampleTypeTraits>(
+        frames_filled, reinterpret_cast<float*>(audio_data));
 
     // Release the buffer space acquired in the GetBuffer() call.
     // Render silence if we were not able to fill up the buffer totally.