Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc

 /*
  *  Copyright (c) 2015 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 29 matching lines @@
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
   RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
   size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
   *speech_type = ConvertSpeechType(1);
   return static_cast<int>(ret);
 }
 
 std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
     rtc::Buffer&& payload,
-    uint32_t timestamp,
-    bool is_primary) {
+    uint32_t timestamp) {
   const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
   return LegacyEncodedAudioFrame::SplitBySamples(
-      this, std::move(payload), timestamp, is_primary,
-      samples_per_ms * 2 * num_channels_, samples_per_ms);
+      this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
+      samples_per_ms);
 }
 
 int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
                                        size_t encoded_len) const {
   // Two encoded byte per sample per channel.
   return static_cast<int>(encoded_len / (2 * Channels()));
 }
 
 }  // namespace webrtc