Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/audio_decoder.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 #include <assert.h>
 #include <memory>
 #include <utility>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/sanitizer.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
 AudioDecoder::ParseResult::ParseResult() = default;
 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
-                                       bool primary,
+                                       uint8_t priority,
                                        std::unique_ptr<EncodedAudioFrame> frame)
-    : timestamp(timestamp), primary(primary), frame(std::move(frame)) {}
+    : timestamp(timestamp), priority(priority), frame(std::move(frame)) {}
 
 AudioDecoder::ParseResult::~ParseResult() = default;
 
 AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
     ParseResult&& b) = default;
 
 std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
     rtc::Buffer* payload,
-    uint32_t timestamp,
-    bool is_primary) {
+    uint32_t timestamp) {
   std::vector<ParseResult> results;
   std::unique_ptr<EncodedAudioFrame> frame(
-      new LegacyEncodedAudioFrame(this, payload, is_primary));
-  results.emplace_back(timestamp, is_primary, std::move(frame));
+      new LegacyEncodedAudioFrame(this, payload));
+  results.emplace_back(timestamp, 0, std::move(frame));
   return results;
 }
 
 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
                          int sample_rate_hz, size_t max_decoded_bytes,
                          int16_t* decoded, SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
   rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
   int duration = PacketDuration(encoded, encoded_len);
   if (duration >= 0 &&
@@ skipping 64 matching lines @@
       return kSpeech;
     case 2:
       return kComfortNoise;
     default:
       assert(false);
       return kSpeech;
   }
 }
 
 }  // namespace webrtc