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AudioEncoder subclass for iSACfix

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This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-12-16 12:49:37 +00:00
parent 0198933b3d
commit 88bdec8c3a
12 changed files with 586 additions and 369 deletions
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108
webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h Normal file
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@ -0,0 +1,108 @@
/*
* 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_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
namespace webrtc {
struct IsacFix {
typedef ISACFIX_MainStruct instance_type;
static const bool has_32kHz = false;
static const uint16_t kFixSampleRate = 16000;
static inline int16_t Control(instance_type* inst,
int32_t rate,
int16_t framesize) {
return WebRtcIsacfix_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 WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
enforce_frame_size);
}
static inline int16_t Create(instance_type** inst) {
return WebRtcIsacfix_Create(inst);
}
static inline int16_t Decode(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
}
static inline int16_t DecodePlc(instance_type* inst,
int16_t* decoded,
int16_t num_lost_frames) {
return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
}
static inline int16_t DecodeRcu(instance_type* inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speech_type) {
// iSACfix has no DecodeRcu; just call the normal Decode.
return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
}
static inline int16_t DecoderInit(instance_type* inst) {
return WebRtcIsacfix_DecoderInit(inst);
}
static inline int16_t Encode(instance_type* inst,
const int16_t* speech_in,
uint8_t* encoded) {
return WebRtcIsacfix_Encode(inst, speech_in, encoded);
}
static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
return WebRtcIsacfix_EncoderInit(inst, coding_mode);
}
static inline uint16_t EncSampRate(instance_type* inst) {
return kFixSampleRate;
}
static inline int16_t Free(instance_type* inst) {
return WebRtcIsacfix_Free(inst);
}
static inline int16_t GetErrorCode(instance_type* inst) {
return WebRtcIsacfix_GetErrorCode(inst);
}
static inline int16_t GetNewFrameLen(instance_type* inst) {
return WebRtcIsacfix_GetNewFrameLen(inst);
}
static inline int16_t SetDecSampRate(instance_type* inst,
uint16_t sample_rate_hz) {
DCHECK_EQ(sample_rate_hz, kFixSampleRate);
return 0;
}
static inline int16_t SetEncSampRate(instance_type* inst,
uint16_t sample_rate_hz) {
DCHECK_EQ(sample_rate_hz, kFixSampleRate);
return 0;
}
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 WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size,
rtp_seq_number, send_ts, arr_ts);
}
};
typedef AudioEncoderDecoderIsacT<IsacFix> AudioEncoderDecoderIsacFix;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_

23
webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc Normal file
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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.
*/
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
namespace webrtc {
const uint16_t IsacFix::kFixSampleRate;
// Explicit instantiation of AudioEncoderDecoderIsacT<IsacFix>, a.k.a.
// AudioEncoderDecoderIsacFix.
template class AudioEncoderDecoderIsacT<IsacFix>;
} // namespace webrtc

4
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
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@ -26,10 +26,14 @@
],
},
'sources': [
'../../audio_encoder_isac_t.h',
'../../audio_encoder_isac_t_impl.h',
'../interface/audio_encoder_isacfix.h',
'../interface/isacfix.h',
'arith_routines.c',
'arith_routines_hist.c',
'arith_routines_logist.c',
'audio_encoder_isacfix.cc',
'bandwidth_estimator.c',
'decode.c',
'decode_bwe.c',