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
 // and implementation of this class.
 class AudioDecoder {
  public:
   enum SpeechType {
     kSpeech = 1,
     kComfortNoise = 2
   };
 
   // Used by PacketDuration below. Save the value -1 for errors.
   enum { kNotImplemented = -2 };
 
   AudioDecoder() = default;
   virtual ~AudioDecoder() = default;
 
   class EncodedAudioFrame {

[kwiberg-webrtc, 2016/09/15 13:01:25]

You don't have to change this, but to me this name doesn't imply that it's a
decoder. Maybe AudioFrameDecoder? AudioFrameWithDecoder? SelfDecodingAudioFrame?
:-)

[ossu, 2016/09/15 14:41:05]

I'd argue that it isn't a decoder, but a frame that can be decoded. It still
relies on having an underlying decoder (which potentially has state), which is
why we (now) promise in the ParsePayload comment that the AudioDecoder object
will outlive every frame from that encoder.

[kwiberg-webrtc, 2016/09/16 00:48:07]

Acknowledged.

    public:
     struct DecodeResult {
       size_t num_decoded_samples;
       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|.
     // Returns rtc::Optional containing the total number of samples across all
     // channels, as well as whether the decoder produced comfort noise or
     // speech. Decode must only be called once per frame object.

[kwiberg-webrtc, 2016/09/15 13:01:24]

Even more specifically: 0 or 1 times.

[ossu, 2016/09/15 14:41:05]

Took care of that in an update to a previous CL.

[kwiberg-webrtc, 2016/09/16 00:48:07]

Acknowledged.

     virtual rtc::Optional<DecodeResult> Decode(
         rtc::ArrayView<int16_t> decoded) const = 0;
   };
 
   struct ParseResult {
     ParseResult();
     ParseResult(uint32_t timestamp,
                 bool primary,
                 std::unique_ptr<EncodedAudioFrame> frame);
     ParseResult(ParseResult&& b);
@@ skipping 96 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_