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Decoder for multistream Opus.

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See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.

This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"

Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
This commit is contained in:
Alex Loiko
2019-04-08 17:19:41 +02:00
committed by Commit Bot
parent e9d2b4efdd
commit e5b94160b5
20 changed files with 918 additions and 96 deletions
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54
modules/audio_coding/codecs/opus/audio_decoder_opus.cc
View File

@ -15,64 +15,14 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
class OpusFrame : public AudioDecoder::EncodedAudioFrame {
public:
OpusFrame(AudioDecoderOpusImpl* 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);
}
bool IsDtxPacket() const override { return payload_.size() <= 2; }
absl::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 absl::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
private:
AudioDecoderOpusImpl* const decoder_;
const rtc::Buffer payload_;
const bool is_primary_payload_;
};
} // namespace
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
: channels_(num_channels) {
RTC_DCHECK(num_channels == 1 || num_channels == 2 || num_channels == 4 ||
num_channels == 6 || num_channels == 8);
RTC_DCHECK(num_channels == 1 || num_channels == 2);
const int error = WebRtcOpus_DecoderCreate(&dec_state_, channels_);
RTC_DCHECK(error == 0);
WebRtcOpus_DecoderInit(dec_state_);