Add an initial revision of an audio server.

services/media/audio/platform/generic/mixer.h

Index: services/media/audio/platform/generic/mixer.h
diff --git a/services/media/audio/platform/generic/mixer.h b/services/media/audio/platform/generic/mixer.h
new file mode 100644
index 0000000000000000000000000000000000000000..f9972b264a631b346127a37d93ba514c69d8464b
--- /dev/null
+++ b/services/media/audio/platform/generic/mixer.h
@@ -0,0 +1,108 @@
+// Copyright 2015 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.
+
+#ifndef SERVICES_MEDIA_AUDIO_PLATFORM_GENERIC_MIXER_H_
+#define SERVICES_MEDIA_AUDIO_PLATFORM_GENERIC_MIXER_H_
+
+#include <memory>
+
+#include "mojo/services/media/common/interfaces/media_types.mojom.h"
+
+namespace mojo {
+namespace media {
+namespace audio {
+
+class Mixer;
+using MixerPtr = std::unique_ptr<Mixer>;
+
+class Mixer {
+ public:
+  virtual ~Mixer();
+
+  // Select
+  //
+  // Select an appropriate instance of a mixer based on the properties of the
+  // source and destination formats.
+  //
+  // TODO(johngro): Come back here and add a way to indicate user preference
+  // where appropriate.  For example, where we might chose a linear
+  // interpolation sampler, the user may actually prefer cubic interpolation, or
+  // perhaps just a point sampler.
+  static MixerPtr Select(const LpcmMediaTypeDetailsPtr& src_format,

[jeffbrown, 2015/11/04 23:43:34]

This is looking much cleaner than the first patch.  Nice.

[johngro, 2015/11/06 02:20:26]

Acknowledged.

+                         const LpcmMediaTypeDetailsPtr& dst_format);
+
+  // Mix
+  //
+  // Perform a mixing operation from the source buffer into the destination
+  // buffer.
+  //
+  // @param dst
+  // The pointer to the destination buffer into which frames will be mixed.
+  //
+  // @param dst_frames
+  // The total number of frames of audio which comprise the destination buffer.
+  //
+  // @param dst_offset
+  // The offset (in destination frames) at which we should start to mix
+  // destination frames.
+  //
+  // @param src
+  // The pointer the the source buffer containing the frames to be mixed into
+  // the destination buffer.
+  //
+  // @param frac_src_frames
+  // The total number of fractional track frames contained by the source buffer.
+  //
+  // @param frac_src_offset
+  // A pointer to the offset (expressed in fractional track frames) at which the
+  // first frame to be mixed with the destination buffer should be sampled.
+  // When Mix has finished, frac_src_offset will be updated to indicate the
+  // offset of the sampling position of the next frame to be mixed with the
+  // output buffer.
+  //
+  // @param frac_step_size
+  // How much to increment the fractional sampling position for each output

[jeffbrown, 2015/11/04 23:43:34]

Is this actually a fraction?  Like a fixed point value of some kind?

[johngro, 2015/11/06 02:20:26]

yes, that is exactly what it is.

+  // frame produced.
+  //
+  // TODO(johngro): Right now, this number may have some amount of rounding
+  // error which will accumulate as sampling position error as we produce more
+  // output samples for a single call to Mix.  This error will reset when we
+  // swtich to the next source buffer, but could (in theory) be the source of
+  // distortion.  If this becomes a problem, we should consider switching to
+  // some form of (N,M) stepping system where we count by frac_step_size for N
+  // output samples, then frac_step_size+1 for M samples, etc...
+  //
+  // @param accumulate
+  // When true, the mixer will accumulate into the destination buffer (read,
+  // sum, clip, write-back).  When false, the mixer will simply replace the
+  // destination buffer with its output.
+  //
+  // @return True if the mixer is finished with this source data and will not
+  // need it in the future.  False if the mixer has not consumed the entire
+  // source buffer and will need more of it in the future.
+  virtual bool Mix(void*       dst,
+                   uint32_t    dst_frames,
+                   uint32_t*   dst_offset,
+                   const void* src,
+                   uint32_t    frac_src_frames,
+                   uint32_t*   frac_src_offset,
+                   uint32_t    frac_step_size,
+                   bool        accumulate) = 0;
+
+  // Reset
+  //
+  // Reset the internal state of the mixer.  Will be called every time there is
+  // a discontinuity in the source stream.  Mixer implementations should reset
+  // anything related to their internal filter state.
+  virtual void Reset() {}
+
+ protected:
+  Mixer();
+};
+
+}  // namespace audio
+}  // namespace media
+}  // namespace mojo
+
+#endif  // SERVICES_MEDIA_AUDIO_PLATFORM_GENERIC_MIXER_H_