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.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 
 #include <memory>
 #include <vector>
-
-#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 30 matching lines @@
     // 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 may be called at most 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,
+                int priority,
                 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.
+    // The highest priority is zero - negative values are not allowed.
+    int priority;
     std::unique_ptr<EncodedAudioFrame> frame;
   };
 
   // Let the decoder parse this payload and prepare zero or more decodable
   // frames. Each frame must be between 10 ms and 120 ms long. The caller must
   // ensure that the AudioDecoder object outlives any frame objects returned by
   // this call. The decoder is free to swap or move the data from the |payload|
   // buffer. |timestamp| is the input timestamp, in samples, corresponding to
   // the start of the payload.
   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 69 matching lines @@
                                       size_t encoded_len,
                                       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_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_