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Move AudioDecoderIsac* to its own files

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Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1339253003 .

Cr-Commit-Position: refs/heads/master@{#10018}
This commit is contained in:
Karl Wiberg
2015-09-22 19:31:40 +02:00
parent 7083e119e8
commit 7404368998
20 changed files with 505 additions and 333 deletions
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22
webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h Normal file
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@ -0,0 +1,22 @@
/*
* 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_MAIN_INTERFACE_AUDIO_DECODER_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_DECODER_ISAC_H_
#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
namespace webrtc {
using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_

104
webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
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@ -11,114 +11,12 @@
#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/isac/audio_encoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.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,
int framesize) {
return WebRtcIsac_Control(inst, rate, framesize);
}
static inline int16_t ControlBwe(instance_type* inst,
int32_t rate_bps,
int 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 int DecodeInternal(instance_type* inst,
const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speech_type) {
return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
}
static inline size_t DecodePlc(instance_type* inst,
int16_t* decoded,
size_t num_lost_frames) {
return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
}
static inline void DecoderInit(instance_type* inst) {
WebRtcIsac_DecoderInit(inst);
}
static inline int 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 void GetBandwidthInfo(instance_type* inst,
IsacBandwidthInfo* bwinfo) {
WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
}
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 void SetBandwidthInfo(instance_type* inst,
const IsacBandwidthInfo* bwinfo) {
WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
}
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 void SetEncSampRateInDecoder(instance_type* inst,
uint16_t sample_rate_hz) {
WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
}
static inline void SetInitialBweBottleneck(
instance_type* inst,
int bottleneck_bits_per_second) {
WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
}
static inline int16_t UpdateBwEstimate(instance_type* inst,
const uint8_t* encoded,
size_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);
}
};
using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_