webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
Niels Möller 48b32b748e Delete support for enabling adaptive isac mode
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-20 10:41:09 +00:00

59 lines
1.9 KiB
C++

/*
* 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_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/scoped_refptr.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
template <typename T>
class AudioDecoderIsacT final : public AudioDecoder {
public:
struct Config {
bool IsOk() const;
int sample_rate_hz = 16000;
};
explicit AudioDecoderIsacT(const Config& config);
virtual ~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_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_