Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 42abd0a0d604d0875d614f39748687a548ace19e..b6d8a3a1db914eeacb08f8229f3e52d53bf8bab9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -10,10 +10,60 @@
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 
+#include <utility>
+
 #include "webrtc/base/checks.h"
 
 namespace webrtc {
 
+namespace {
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoderOpus* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  rtc::Optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    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);
+    }
+
+    if (ret < 0)
+      return rtc::Optional<DecodeResult>();
+
+    return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
+  }
+
+ private:
+  AudioDecoderOpus* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace
+
 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
     : channels_(num_channels) {
   RTC_DCHECK(num_channels == 1 || num_channels == 2);
@@ -25,6 +75,26 @@ AudioDecoderOpus::~AudioDecoderOpus() {
   WebRtcOpus_DecoderFree(dec_state_);
 }
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpus::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
 int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,