Chromium Code Reviews| 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..a31c9b07aed1f69e06762b04422a7591215b0ad4 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 +10,61 @@ |
| #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), is_primary_payload_(is_primary_payload) { |
| + using std::swap; |
| + swap(this->payload_, payload); |
|
hlundin-webrtc
2016/09/15 09:33:30
std::move?
ossu
2016/09/15 12:22:53
Acknowledged.
|
| + } |
| + |
| + 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()); |
| + } |
|
kwiberg-webrtc
2016/09/19 11:07:49
const:
const int ret = (is_primary_payload_ ? dec
ossu
2016/09/19 11:41:01
Oh, god, no! I'll be surprised if it works but eve
kwiberg-webrtc
2016/09/19 11:55:16
Hmm, you're right, I don't think it will work. And
hlundin-webrtc
2016/09/19 12:19:13
I'm with ossu: "Oh, God, no!"
ossu
2016/09/19 14:07:33
?: is fine but it's easy to end up with a big, con
|
| + 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); |
| + } |
|
kwiberg-webrtc
2016/09/19 11:07:49
The same ?: construction can be used to make ret c
kwiberg-webrtc
2016/09/19 11:55:16
No, it won't work for the same reason as above.
|
| + |
| + if (ret < 0) |
| + return rtc::Optional<DecodeResult>(); |
| + |
| + return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type}); |
| + } |
| + |
| + private: |
| + AudioDecoderOpus* decoder_; |
|
hlundin-webrtc
2016/09/15 09:33:30
const, const, const
ossu
2016/09/15 12:22:53
Acknowledged. Acknowledged. Acknowledged.
ossu
2016/09/19 11:41:01
Acknowledged.
kwiberg-webrtc
2016/09/19 11:55:16
Acknowledged.
hlundin-webrtc
2016/09/19 12:19:13
Roger that.
kwiberg-webrtc
2016/09/20 09:14:45
Roger, Roger. What's our vector, Victor?
|
| + rtc::Buffer payload_; |
| + bool is_primary_payload_; |
|
kwiberg-webrtc
2016/09/19 11:07:49
Not sure if this is what Henrik was pointing out,
hlundin-webrtc
2016/09/19 12:19:13
That is why I put three "const" in one comment. I
ossu
2016/09/19 14:07:33
I was only able to get two consts into the decoder
|
| +}; |
| + |
| +} // namespace |
| + |
| AudioDecoderOpus::AudioDecoderOpus(size_t num_channels) |
| : channels_(num_channels) { |
| RTC_DCHECK(num_channels == 1 || num_channels == 2); |
| @@ -25,6 +76,25 @@ 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 = |
|
hlundin-webrtc
2016/09/15 09:33:29
What if PacketDurationRedundant returns an error (
ossu
2016/09/15 12:22:53
I can't see that it's able to: it will call WebRtc
hlundin-webrtc
2016/09/16 07:51:27
But the method comment for AudioDecoder::PacketDur
ossu
2016/09/19 11:41:01
But it is only used within this implementation, i.
hlundin-webrtc
2016/09/19 12:19:13
You are right. Since this is an internal affair of
ossu
2016/09/19 14:07:33
Acknowledged.
|
| + PacketDurationRedundant(payload->data(), payload->size()); |
| + 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, |