| 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 a611ecbe16f76222a0130142c66fd7cc2a929655..f258c1e07e27fb3df4612f261d64b9e8e59d06aa 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());
|
| - }
|
| + const 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);
|
| - }
|
| + const int ret = decoder_->Decode(
|
| + 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, std::move(payload), is_primary));
|
| - results.emplace_back(timestamp, is_primary, std::move(frame));
|
| + new LegacyEncodedAudioFrame(decoder, std::move(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
|
| @@ -90,10 +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, std::move(new_payload),
|
| - is_primary));
|
| - results.emplace_back(timestamp + timestamp_offset, is_primary,
|
| - std::move(frame));
|
| + new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
|
| + results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
|
| }
|
| }
|
|
|
|
|