Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 
+#include <utility>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
 AudioDecoderIlbc::AudioDecoderIlbc() {
   WebRtcIlbcfix_DecoderCreate(&dec_state_);
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
@@ skipping 23 matching lines @@
 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
   return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
 void AudioDecoderIlbc::Reset() {
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
 std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload(
     rtc::Buffer&& payload,
-    uint32_t timestamp,
-    bool is_primary) {
+    uint32_t timestamp) {
   std::vector<ParseResult> results;
   size_t bytes_per_frame;
   int timestamps_per_frame;
   if (payload.size() >= 950) {
     LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Payload too large";
     return results;
   }
   if (payload.size() % 38 == 0) {
     // 20 ms frames.
     bytes_per_frame = 38;
     timestamps_per_frame = 160;
   } else if (payload.size() % 50 == 0) {
     // 30 ms frames.
     bytes_per_frame = 50;
     timestamps_per_frame = 240;
   } else {
     LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload";
     return results;
   }
 
   RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame);
   if (payload.size() == bytes_per_frame) {
     std::unique_ptr<EncodedAudioFrame> frame(
-        new LegacyEncodedAudioFrame(this, std::move(payload), is_primary));
-    results.emplace_back(timestamp, is_primary, std::move(frame));
+        new LegacyEncodedAudioFrame(this, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
   } else {
     for (size_t byte_offset = 0, timestamp_offset = 0;
          byte_offset < payload.size();
          byte_offset += bytes_per_frame,
              timestamp_offset += timestamps_per_frame) {
       rtc::Buffer new_payload(payload.data() + byte_offset, bytes_per_frame);
-      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
-          this, std::move(new_payload), is_primary));
-      results.emplace_back(timestamp + timestamp_offset, is_primary,
-                           std::move(frame));
+      std::unique_ptr<EncodedAudioFrame> frame(
+          new LegacyEncodedAudioFrame(this, std::move(new_payload)));

[kwiberg-webrtc, 2016/09/20 09:14:45]

Arguably, new_payload doesn't have to be a named variable anymore. Eliminate it
if you feel like it.

[ossu, 2016/09/20 13:51:56]

Acknowledged.

+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
     }
   }
 
   return results;
 }
 
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }
 
 size_t AudioDecoderIlbc::Channels() const {
   return 1;
 }
 
 }  // namespace webrtc