Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/audio_decoder.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
 
-#include <stdlib.h>  // NULL
+#include <memory>
+#include <vector>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // This is the interface class for decoders in NetEQ. Each codec type will have
@@ skipping 18 matching lines @@
       SpeechType speech_type;
     };
 
     virtual ~EncodedAudioFrame() = default;
 
     // Returns the duration in samples-per-channel of this audio frame.
     // If no duration can be ascertained, returns zero.
     virtual size_t Duration() const = 0;
 
     // Decodes this frame of audio and writes the result in |decoded|.
     // |decoded| will be large enough for 120 ms of audio at the decoder's

[hlundin-webrtc, 2016/09/16 07:35:20]

"|decoded| will be large enough"
This sounds like the Decode nethod will sort that for you. Should you not say
that is *must* be large enough?

[kwiberg-webrtc, 2016/09/16 07:56:33]

Yeah, it's always troublesome to document interfaces, since the docs have to
work for both sides. So yes, I agree that "must" is a better choice of words.

[ossu, 2016/09/16 11:46:00]

Hmm. It mustn't really, though. Since we have a Duration call, the reasonable
request would be to have the caller make sure decoded is large enough for the
frame based on the value returned by Decoded. That would let us keep the
specifics of frame sizes in the comment describing ParsePayload. I've updated
this comment in an earlier CL.

[kwiberg-webrtc, 2016/09/16 12:07:39]

Acknowledged.

     // sample rate. On success, returns an rtc::Optional containing the total
     // number of samples across all channels, as well as whether the decoder
     // produced comfort noise or speech. On failure, returns an empty
     // rtc::Optional. Decode must be called at most once per frame object.
     virtual rtc::Optional<DecodeResult> Decode(
         rtc::ArrayView<int16_t> decoded) const = 0;
   };
 
   struct ParseResult {
     ParseResult();
@@ skipping 103 matching lines @@
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_