Implement renderer AudioDevice API for low-latency audio output

BUG=none
TEST=none
(I tested locally for Mac OS X, but we'll need further testing for Windows and Linux as we add real-time threads, etc.)
Review URL: http://codereview.chromium.org/6002005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@71521 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 31 insertions, 0 deletions

diff --git a/media/audio/audio_util.cc b/media/audio/audio_util.cc
index 6d80c87..ed84bc1 100644
--- a/media/audio/audio_util.cc
+++ b/media/audio/audio_util.cc
@@ -206,4 +206,23 @@ bool DeinterleaveAudioChannel(void* source,
   return false;
 }
 
+void InterleaveFloatToInt16(const std::vector<float*>& source,
+                            int16* destination,
+                            size_t number_of_frames) {
+  const float kScale = 32768.0f;
+  int channels = source.size();
+  for (int i = 0; i < channels; ++i) {
+    float* channel_data = source[i];
+    for (size_t j = 0; j < number_of_frames; ++j) {
+      float sample = kScale * channel_data[j];
+      if (sample < -32768.0)
+        sample = -32768.0;
+      else if (sample > 32767.0)
+        sample = 32767.0;
+
+      destination[j * channels + i] = static_cast<int16>(sample);
+    }
+  }
+}
+
 }  // namespace media
diff --git a/media/audio/audio_util.h b/media/audio/audio_util.h
index 669b4fa..a6ba75f 100644
--- a/media/audio/audio_util.h
+++ b/media/audio/audio_util.h
@@ -5,6 +5,8 @@
 #ifndef MEDIA_AUDIO_AUDIO_UTIL_H_
 #define MEDIA_AUDIO_AUDIO_UTIL_H_
 
+#include <vector>
+
 #include "base/basictypes.h"
 
 namespace media {
@@ -62,6 +64,16 @@ bool DeinterleaveAudioChannel(void* source,
                               int bytes_per_sample,
                               size_t number_of_frames);
 
+// InterleaveFloatToInt16 scales, clips, and interleaves the planar
+// floating-point audio contained in |source| to the int16 |destination|.
+// The floating-point data is in a canonical range of -1.0 -> +1.0.
+// The size of the |source| vector determines the number of channels.
+// The |destination| buffer is assumed to be large enough to hold the
+// result. Thus it must be at least size: number_of_frames * source.size()
+void InterleaveFloatToInt16(const std::vector<float*>& source,
+                            int16* destination,
+                            size_t number_of_frames);
+
 }  // namespace media
 
 #endif  // MEDIA_AUDIO_AUDIO_UTIL_H_