Moved codec-specific audio packet splitting into decoders.

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 {
 
-namespace {
-class LegacyFrame : public AudioDecoder::EncodedAudioFrame {
- public:
-  LegacyFrame(AudioDecoder* 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:
-  AudioDecoder* const decoder_;
-  const rtc::Buffer payload_;
-  const bool is_primary_payload_;
-};
-}
-
 AudioDecoder::ParseResult::ParseResult() = default;
 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
                                        bool primary,
                                        std::unique_ptr<EncodedAudioFrame> frame)
     : timestamp(timestamp), primary(primary), 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) {
   std::vector<ParseResult> results;
   std::unique_ptr<EncodedAudioFrame> frame(
-      new LegacyFrame(this, payload, is_primary));
+      new LegacyEncodedAudioFrame(this, payload, is_primary));
   results.emplace_back(timestamp, is_primary, std::move(frame));
   return results;

[kwiberg-webrtc, 2016/09/15 13:01:24]

Won't a construction along these lines work?

  std::unique_ptr<EncodedAudioFrame> frame(
      new LegacyEncodedAudioFrame(this, payload, is_primary));
  return {{timestamp, is_primary, std::move(frame)}};

[ossu, 2016/09/15 14:41:05]

Ideally that _should_ work but I can't get it to. I wonder if the initializer
list constructor is somehow disabled, seeing as Chromium doesn't allow
initializer lists. :/

[kwiberg-webrtc, 2016/09/16 00:48:07]

No, they're allowed now. Could it be that we're using a language whose syntax is
more Byzantine than we're comfortable with?

 }
 
 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 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@@ skipping 63 matching lines @@
       return kSpeech;
     case 2:
       return kComfortNoise;
     default:
       assert(false);
       return kSpeech;
   }
 }
 
 }  // namespace webrtc