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/*
* Copyright (c) 2014 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_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
namespace webrtc {
struct IsacFloat {
typedef ISACStruct instance_type;
static const bool has_swb = true;
static inline int16_t Control(instance_type* inst,
int32_t rate,
int16_t framesize) {
return WebRtcIsac_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
int16_t frame_size_ms,
int16_t enforce_frame_size) {
return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
enforce_frame_size);
}
static inline int16_t Create(instance_type** inst) {
return WebRtcIsac_Create(inst);
}
static inline int16_t DecodeInternal(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
}
static inline int16_t DecodePlc(instance_type* inst,
int16_t* decoded,
int16_t num_lost_frames) {
return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
}
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsac_DecoderInit(inst);
}
static inline int16_t Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
return WebRtcIsac_Encode(inst, speech_in, encoded);
}
static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
return WebRtcIsac_EncoderInit(inst, coding_mode);
}
static inline uint16_t EncSampRate(instance_type* inst) {
return WebRtcIsac_EncSampRate(inst);
}
static inline int16_t Free(instance_type* inst) {
return WebRtcIsac_Free(inst);
}
static inline int16_t GetErrorCode(instance_type* inst) {
return WebRtcIsac_GetErrorCode(inst);
}
static inline int16_t GetNewFrameLen(instance_type* inst) {
return WebRtcIsac_GetNewFrameLen(inst);
}
static inline int16_t SetDecSampRate(instance_type* inst,
uint16_t sample_rate_hz) {
return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
}
static inline int16_t SetEncSampRate(instance_type* inst,
uint16_t sample_rate_hz) {
return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
}
static inline int16_t UpdateBwEstimate(instance_type* inst,
const uint8_t* encoded,
int32_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts,
uint32_t arr_ts) {
return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
rtp_seq_number, send_ts, arr_ts);
}
static inline int16_t SetMaxPayloadSize(instance_type* inst,
int16_t max_payload_size_bytes) {
return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
}
static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
return WebRtcIsac_SetMaxRate(inst, max_bit_rate);
}
};
typedef AudioEncoderDecoderIsacT<IsacFloat> AudioEncoderDecoderIsac;
struct CodecInst;
class AudioEncoderDecoderMutableIsacFloat
: public AudioEncoderMutableImpl<AudioEncoderDecoderIsac,
AudioEncoderDecoderMutableIsac> {
public:
explicit AudioEncoderDecoderMutableIsacFloat(const CodecInst& codec_inst);
void UpdateSettings(const CodecInst& codec_inst) override;
void SetMaxPayloadSize(int max_payload_size_bytes) override;
void SetMaxRate(int max_rate_bps) override;
// From AudioDecoder.
int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) override;
int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) override;
bool HasDecodePlc() const override;
int DecodePlc(int num_frames, int16_t* decoded) override;
int Init() 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 PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const override;
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
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