Chromium Code Reviews| Index: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc |
| diff --git a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc |
| index fd79dc9dd0eb6aa40846020c34d0ecd86564fdff..2edf1a3beadbc12ca2b943ba28c78b79f94649e2 100644 |
| --- a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc |
| +++ b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc |
| @@ -17,37 +17,22 @@ |
| 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>(); |
| @@ -59,7 +44,6 @@ 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()); |
| @@ -70,8 +54,8 @@ std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples( |
| 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 |
| @@ -90,9 +74,8 @@ std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples( |
| 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)); |
| } |
| } |