Moved Opus-specific payload splitting into AudioDecoderOpus.

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.
  */
 
@@ skipping 43 matching lines @@
     // 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.
     virtual rtc::Optional<DecodeResult> Decode(
         rtc::ArrayView<int16_t> decoded) const = 0;
   };
 
   struct ParseResult {
     ParseResult();
     ParseResult(uint32_t timestamp,
-                bool primary,
+                uint8_t priority,

[hlundin-webrtc, 2016/09/15 09:33:29]

"You should not use the unsigned integer types such as uint32_t, unless there is
a valid reason such as representing a bit pattern rather than a number, or you
need defined overflow modulo 2^N. In particular, do not use unsigned types to
say a number will never be negative. Instead, use assertions for this."
https://google.github.io/styleguide/cppguide.html#Integer_Types

[ossu, 2016/09/15 12:22:53]

You are technically correct - the best kind of correct! Perhaps these should
just be ints after all. Seems like a waste of bytes, though. Not that they are
very expensive...

[hlundin-webrtc, 2016/09/16 07:51:27]

Once upon a time, in an office building (not so) far far away, we used to spend
a lot of time tweaking the size of member variables. This code is more int than
uint8_t these days...

[kwiberg-webrtc, 2016/09/19 11:07:49]

Function arguments and local variables are stored in registers most of the time
unless they're live when you call a subroutine, and then (almost) any sort of
integer will occupy a full register.

+1 for making this an int.

[ossu, 2016/09/19 11:41:01]

I've thought about this change as also applying to Packet::Priority, which is
where the size "actually" matters. There having two ints where none can be < 0
and both will be really small is more of a waste, and a hassle. I can have them
like that, though, with CHECKs for < 0, but I think we're throwing away some
expressiveness in the API.

[kwiberg-webrtc, 2016/09/19 11:55:16]

I also thought about this change as applying to Packet::Priority. Why would you
need to CHECK that the values are nonnegative?

[ossu, 2016/09/19 14:07:33]

If we want some priority level to correspond to "normal audio packet from a
non-RED payload or primary RED-sub-block", e.g. for stats.

[kwiberg-webrtc, 2016/09/20 09:14:44]

Ah, yes, for stats I can see why we'd care about the "absolute" priority and not
just the relative priority. Thanks.

                 std::unique_ptr<EncodedAudioFrame> frame);
     ParseResult(ParseResult&& b);
     ~ParseResult();
 
     ParseResult& operator=(ParseResult&& b);
 
     // The timestamp of the frame is in samples per channel.
     uint32_t timestamp;
-    bool primary;
+    // The relative priority of the frame compared to other frames of the same
+    // payload and the same timeframe. A higher value means a lower priority.
+    uint8_t priority;
     std::unique_ptr<EncodedAudioFrame> frame;
   };
 
   // Let the decoder parse this payload and prepare zero or more decodable
   // frames. Each frame must be at most 120 ms long. The decoder is free to swap
   // or move the data from the |payload| buffer.
   virtual std::vector<ParseResult> ParsePayload(rtc::Buffer* payload,
-                                                uint32_t timestamp,
-                                                bool is_primary);
+                                                uint32_t timestamp);
 
   // Decodes |encode_len| bytes from |encoded| and writes the result in
   // |decoded|. The maximum bytes allowed to be written into |decoded| is
   // |max_decoded_bytes|. Returns the total number of samples across all
   // channels. If the decoder produced comfort noise, |speech_type|
   // is set to kComfortNoise, otherwise it is kSpeech. The desired output
   // sample rate is provided in |sample_rate_hz|, which must be valid for the
   // codec at hand.
   int Decode(const uint8_t* encoded,
              size_t encoded_len,
@@ skipping 70 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_