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..48332885e75e412a0d27f2cdc8e45eb3a006c2fd 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -6,6 +6,8 @@
 
 #include <Functiondiscoverykeys_devpkey.h>
 
+#include <climits>
+
 #include "base/command_line.h"
 #include "base/logging.h"
 #include "base/macros.h"
@@ -18,6 +20,7 @@
 #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"
 
@@ -71,8 +74,7 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
       device_role_(device_role),
       share_mode_(GetShareMode()),
       num_written_frames_(0),
-      source_(NULL),
-      audio_bus_(AudioBus::Create(params)) {
+      source_(NULL) {
   DCHECK(manager_);
 
   // The empty string is used to indicate a default device and the
@@ -89,6 +91,15 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
   bool avrt_init = avrt::Initialize();
   DCHECK(avrt_init) << "Failed to load the avrt.dll";
 
+  // New set that appropriate for float output.
+  AudioParameters float_params(
+      params.format(), params.channel_layout(), params.sample_rate(),
+      // Ignore the given bits per sample because we're outputting
+      // floats.
+      sizeof(float) * CHAR_BIT, params.frames_per_buffer());
+
+  audio_bus_ = AudioBus::Create(float_params);
+
   // 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.
@@ -96,27 +107,27 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
   // Begin with the WAVEFORMATEX structure that specifies the basic format.
   WAVEFORMATEX* format = &format_.Format;
   format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
-  format->nChannels = params.channels();
-  format->nSamplesPerSec = params.sample_rate();
-  format->wBitsPerSample = params.bits_per_sample();
+  format->nChannels = float_params.channels();
+  format->nSamplesPerSec = float_params.sample_rate();
+  format->wBitsPerSample = float_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_.Samples.wValidBitsPerSample = float_params.bits_per_sample();
   format_.dwChannelMask = CoreAudioUtil::GetChannelConfig(device_id, eRender);
-  format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+  format_.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
 
   // 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.frames_per_buffer();
-  packet_size_bytes_ = params.GetBytesPerBuffer();
+  packet_size_frames_ = float_params.frames_per_buffer();
+  packet_size_bytes_ = float_params.GetBytesPerBuffer();
   DVLOG(1) << "Number of bytes per audio frame  : " << format->nBlockAlign;
   DVLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
   DVLOG(1) << "Number of bytes per packet       : " << packet_size_bytes_;
   DVLOG(1) << "Number of milliseconds per packet: "
-          << params.GetBufferDuration().InMillisecondsF();
+           << float_params.GetBufferDuration().InMillisecondsF();
 
   // All events are auto-reset events and non-signaled initially.
 
@@ -544,13 +555,9 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     uint32_t num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
     DCHECK_LE(num_filled_bytes, packet_size_bytes_);
 
-    // Note: If this ever changes to output raw float the data must be
-    // clipped and sanitized since it may come from an untrusted
-    // 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.