Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc

 /*
  *  Copyright (c) 2016 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/legacy_encoded_audio_frame.h"
 
 #include <algorithm>
 #include <memory>
 #include <utility>
 
 namespace webrtc {
 
 LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
-                                                 rtc::Buffer* payload,
-                                                 bool is_primary_payload)
-    : decoder_(decoder),
-      payload_(std::move(*payload)),
-      is_primary_payload_(is_primary_payload) {}
+                                                 rtc::Buffer* payload)
+    : decoder_(decoder), payload_(std::move(*payload)) {}
 
 LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
 
 size_t LegacyEncodedAudioFrame::Duration() const {
-  int ret;
-  if (is_primary_payload_) {
-    ret = decoder_->PacketDuration(payload_.data(), payload_.size());
-  } else {
-    ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
-  }
+  int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
   return (ret < 0) ? 0 : static_cast<size_t>(ret);
 }
 
 rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
 LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
   AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
-  int ret;
-  if (is_primary_payload_) {
-    ret = decoder_->Decode(
-        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
-        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
-  } else {
-    ret = decoder_->DecodeRedundant(
-        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
-        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
-  }
+  int ret = decoder_->Decode(

[kwiberg-webrtc, 2016/09/19 11:07:49]

const

[ossu, 2016/09/19 11:41:01]

Acknowledged.

+      payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+      decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
 
   if (ret < 0)
     return rtc::Optional<DecodeResult>();
 
   return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
 }
 
 std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
     AudioDecoder* decoder,
     rtc::Buffer* payload,
     uint32_t timestamp,
-    bool is_primary,
     size_t bytes_per_ms,
     uint32_t timestamps_per_ms) {
   RTC_DCHECK(payload->data());
   std::vector<AudioDecoder::ParseResult> results;
   size_t split_size_bytes = payload->size();
 
   // Find a "chunk size" >= 20 ms and < 40 ms.
   const size_t min_chunk_size = bytes_per_ms * 20;
   if (min_chunk_size >= payload->size()) {
     std::unique_ptr<LegacyEncodedAudioFrame> frame(
-        new LegacyEncodedAudioFrame(decoder, payload, is_primary));
-    results.emplace_back(timestamp, is_primary, std::move(frame));
+        new LegacyEncodedAudioFrame(decoder, payload));
+    results.emplace_back(timestamp, 0, std::move(frame));
   } else {
     // Reduce the split size by half as long as |split_size_bytes| is at least
     // twice the minimum chunk size (so that the resulting size is at least as
     // large as the minimum chunk size).
     while (split_size_bytes >= 2 * min_chunk_size) {
       split_size_bytes >>= 1;
     }
 
     const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
         split_size_bytes * timestamps_per_ms / bytes_per_ms);
     for (size_t byte_offset = 0, timestamp_offset = 0;
          byte_offset < payload->size();
          byte_offset += split_size_bytes,
              timestamp_offset += timestamps_per_chunk) {
       split_size_bytes =
           std::min(split_size_bytes, payload->size() - byte_offset);
       rtc::Buffer new_payload(payload->data() + byte_offset, split_size_bytes);
       std::unique_ptr<LegacyEncodedAudioFrame> frame(
-          new LegacyEncodedAudioFrame(decoder, &new_payload, is_primary));
-      results.emplace_back(timestamp + timestamp_offset, is_primary,
-                           std::move(frame));
+          new LegacyEncodedAudioFrame(decoder, &new_payload));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
     }
   }
 
   return results;
 }
 
 }  // namespace webrtc