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/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include <memory>
#include <utility>
#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 {
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
int sample_rate_hz)
: channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
const int error =
WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
RTC_DCHECK(error == 0);
WebRtcOpus_DecoderInit(dec_state_);
}
AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
WebRtcOpus_DecoderFree(dec_state_);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
if (PacketHasFec(payload.data(), payload.size())) {
const int duration =
PacketDurationRedundant(payload.data(), payload.size());
RTC_DCHECK_GE(duration, 0);
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 AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
int16_t temp_type = 1; // Default is speech.
int ret =
WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
&temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
void AudioDecoderOpusImpl::Reset() {
WebRtcOpus_DecoderInit(dec_state_);
}
int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
}
int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return PacketDuration(encoded, encoded_len);
}
return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
}
bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
int fec;
fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
return (fec == 1);
}
int AudioDecoderOpusImpl::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioDecoderOpusImpl::Channels() const {
return channels_;
}
} // namespace webrtc
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