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 <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 };
 
+  struct PacketSplit {
+    size_t byte_offset;
+    size_t num_bytes;
+    size_t timestamp_offset;
+  };
+
   AudioDecoder() = default;
   virtual ~AudioDecoder() = default;
 
   class Frame {
    public:
     struct DecodeResult {
       size_t num_decoded_samples;
       SpeechType speech_type;
     };
 
@@ skipping 65 matching lines @@
   // Resets the decoder state (empty buffers etc.).
   virtual void Reset() = 0;
 
   // Notifies the decoder of an incoming packet to NetEQ.
   virtual int IncomingPacket(const uint8_t* payload,
                              size_t payload_len,
                              uint16_t rtp_sequence_number,
                              uint32_t rtp_timestamp,
                              uint32_t arrival_timestamp);
 
+  // This method is experimental and should not be overriden by external codecs.
+  // It calculates positions where the packet can be split, and the respective
+  // timestamps of those positions. It is part of the work to separate codec
+  // specific functionality from NetEq and will likely be removed soon in favor
+  // of something much better.
+  virtual std::vector<PacketSplit> SplitPacket(
+      rtc::ArrayView<const uint8_t> payload) const;

[kwiberg-webrtc, 2016/09/12 02:11:01]

If you turned the return value into an output argument, you would (with the
cooperation of the caller) not have to pay for a heap allocation and
deallocation for each call.

[ossu, 2016/09/12 11:26:37]

I've addressed this in the comment on SplitBySamples. See below!

+
   // Returns the last error code from the decoder.
   virtual int ErrorCode();
 
   // Returns the duration in samples-per-channel of the payload in |encoded|
   // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
   // estimate is available, or -1 in case of an error.
   virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
 
   // Returns the duration in samples-per-channel of the redandant payload in
   // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
@@ skipping 28 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_