Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/audio_decoder.h

Index: webrtc/modules/audio_coding/codecs/audio_decoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index b6338d2102a490d7c7272def3a62691daebef6eb..af16095117fe2e7a08f47d045343f8c9a91314e8 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -66,7 +66,7 @@ class AudioDecoder {
   struct ParseResult {
     ParseResult();
     ParseResult(uint32_t timestamp,
-                bool primary,
+                int priority,
                 std::unique_ptr<EncodedAudioFrame> frame);
     ParseResult(ParseResult&& b);
     ~ParseResult();
@@ -75,7 +75,9 @@ class AudioDecoder {
 
     // 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.
+    int priority;
     std::unique_ptr<EncodedAudioFrame> frame;
   };
 
@@ -86,8 +88,7 @@ class AudioDecoder {
   // 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