blob: 66f021a87c16047a7c3c78b2f701b9db13602a9c [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 WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
namespace webrtc {
template <typename T>
class AudioDecoderIsacT final : public AudioDecoder {
public:
explicit AudioDecoderIsacT(int sample_rate_hz);
AudioDecoderIsacT(int sample_rate_hz,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
~AudioDecoderIsacT() override;
bool HasDecodePlc() const override;
size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
void Reset() override;
int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) override;
int ErrorCode() override;
int SampleRateHz() const override;
size_t Channels() const override;
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
typename T::instance_type* isac_state_;
int sample_rate_hz_;
rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_