Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
index 8d4a2bc175ba624bfc518adaf4fcebe9eea8d795..468af72894357ef75559add84781a3b522e117cf 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
@@ -19,6 +19,78 @@ |
namespace webrtc { |
+namespace { |
+class LegacyFrame : public AudioDecoder::Frame { |
+ public: |
+ LegacyFrame(AudioDecoder* decoder, |
+ rtc::Buffer* payload, |
+ bool is_primary_payload) |
+ : decoder_(decoder), is_primary_payload_(is_primary_payload) { |
+ using std::swap; |
hlundin-webrtc
2016/09/09 12:11:49
Why the using statement?
kwiberg-webrtc
2016/09/10 07:34:59
It's the standard idiom for swapping stuff. See e.
hlundin-webrtc
2016/09/12 08:07:11
Acknowledged.
|
+ swap(this->payload_, *payload); |
hlundin-webrtc
2016/09/09 12:11:49
#include what you use.
kwiberg-webrtc
2016/09/10 07:34:59
+1
ossu
2016/09/12 10:31:36
Acknowledged.
|
+ } |
kwiberg-webrtc
2016/09/10 07:34:59
Why not move *payload to payload_ instead of swapp
ossu
2016/09/12 10:31:36
Hmm. I was thinking of doing this to allow buffer
|
+ |
+ 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* decoder_; |
hlundin-webrtc
2016/09/09 12:11:49
const
|
+ rtc::Buffer payload_; |
+ bool is_primary_payload_; |
hlundin-webrtc
2016/09/09 12:11:49
const
kwiberg-webrtc
2016/09/10 07:34:59
All three could be const, right?
ossu
2016/09/12 10:31:36
Acknowledged.
|
+}; |
+} |
+ |
+AudioDecoder::ParseResult::ParseResult() = default; |
+AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; |
+AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, |
+ bool primary, |
+ std::unique_ptr<Frame> 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<Frame> frame( |
+ new LegacyFrame(this, payload, is_primary)); |
+ results.emplace_back(timestamp, is_primary, std::move(frame)); |
+ return results; |
+} |
+ |
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) { |