blob: b2783c67076fc470aa1aa4b9b8f15b3638eaa122 [file] [log] [blame]
/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
#include "rtc_base/checks.h"
namespace webrtc {
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(int sample_rate_hz)
: AudioDecoderIsacT(sample_rate_hz, nullptr) {}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(
int sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
: sample_rate_hz_(sample_rate_hz), bwinfo_(bwinfo) {
RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_CHECK_EQ(0, T::Create(&isac_state_));
T::DecoderInit(isac_state_);
if (bwinfo_) {
IsacBandwidthInfo bi;
T::GetBandwidthInfo(isac_state_, &bi);
bwinfo_->Set(bi);
}
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_));
}
template <typename T>
AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
RTC_CHECK_EQ(0, T::Free(isac_state_));
}
template <typename T>
int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
int ret =
T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
template <typename T>
bool AudioDecoderIsacT<T>::HasDecodePlc() const {
return false;
}
template <typename T>
size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
return T::DecodePlc(isac_state_, decoded, num_frames);
}
template <typename T>
void AudioDecoderIsacT<T>::Reset() {
T::DecoderInit(isac_state_);
}
template <typename T>
int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
rtp_sequence_number, rtp_timestamp,
arrival_timestamp);
if (bwinfo_) {
IsacBandwidthInfo bwinfo;
T::GetBandwidthInfo(isac_state_, &bwinfo);
bwinfo_->Set(bwinfo);
}
return ret;
}
template <typename T>
int AudioDecoderIsacT<T>::ErrorCode() {
return T::GetErrorCode(isac_state_);
}
template <typename T>
int AudioDecoderIsacT<T>::SampleRateHz() const {
return sample_rate_hz_;
}
template <typename T>
size_t AudioDecoderIsacT<T>::Channels() const {
return 1;
}
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_