FFmpeg: libavcodec/adxenc.c Source File

FFmpeg
adxenc.c
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1 /*
2  * ADX ADPCM codecs
3  * Copyright (c) 2001,2003 BERO
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "avcodec.h"
23 #include "adx.h"
24 #include "bytestream.h"
25 #include "internal.h"
26 #include "put_bits.h"
27 
28 /**
29  * @file
30  * SEGA CRI adx codecs.
31  *
32  * Reference documents:
33  * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
34  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
35  */
36 
37  static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
38  ADXChannelState *prev, int channels)
39 {
40  PutBitContext pb;
41  int scale;
42  int i, j;
43  int s0, s1, s2, d;
44  int max = 0;
45  int min = 0;
46 
47  s1 = prev->s1;
48  s2 = prev->s2;
49  for (i = 0, j = 0; j < 32; i += channels, j++) {
50  s0 = wav[i];
51  d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
52  if (max < d)
53  max = d;
54  if (min > d)
55  min = d;
56  s2 = s1;
57  s1 = s0;
58  }
59 
60  if (max == 0 && min == 0) {
61  prev->s1 = s1;
62  prev->s2 = s2;
63  memset(adx, 0, BLOCK_SIZE);
64  return;
65  }
66 
67  if (max / 7 > -min / 8)
68  scale = max / 7;
69  else
70  scale = -min / 8;
71 
72  if (scale == 0)
73  scale = 1;
74 
75  AV_WB16(adx, scale);
76 
77  init_put_bits(&pb, adx + 2, 16);
78 
79  s1 = prev->s1;
80  s2 = prev->s2;
81  for (i = 0, j = 0; j < 32; i += channels, j++) {
82  d = ((wav[i] << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
83 
84  d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
85 
86  put_sbits(&pb, 4, d);
87 
88  s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
89  s2 = s1;
90  s1 = s0;
91  }
92  prev->s1 = s1;
93  prev->s2 = s2;
94 
95  flush_put_bits(&pb);
96 }
97 
98  #define HEADER_SIZE 36
99 
100  static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
101 {
102  ADXContext *c = avctx->priv_data;
103 
104  bytestream_put_be16(&buf, 0x8000); /* header signature */
105  bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
106  bytestream_put_byte(&buf, 3); /* encoding */
107  bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
108  bytestream_put_byte(&buf, 4); /* sample size */
109  bytestream_put_byte(&buf, avctx->channels); /* channels */
110  bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
111  bytestream_put_be32(&buf, 0); /* total sample count */
112  bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
113  bytestream_put_byte(&buf, 3); /* version */
114  bytestream_put_byte(&buf, 0); /* flags */
115  bytestream_put_be32(&buf, 0); /* unknown */
116  bytestream_put_be32(&buf, 0); /* loop enabled */
117  bytestream_put_be16(&buf, 0); /* padding */
118  bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
119 
120  return HEADER_SIZE;
121 }
122 
123  static av_cold int adx_encode_init(AVCodecContext *avctx)
124 {
125  ADXContext *c = avctx->priv_data;
126 
127  if (avctx->channels > 2) {
128  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
129  return AVERROR(EINVAL);
130  }
131  avctx->frame_size = BLOCK_SAMPLES;
132 
133  /* the cutoff can be adjusted, but this seems to work pretty well */
134  c->cutoff = 500;
135  ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
136 
137  return 0;
138 }
139 
140  static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
141  const AVFrame *frame, int *got_packet_ptr)
142 {
143  ADXContext *c = avctx->priv_data;
144  const int16_t *samples = (const int16_t *)frame->data[0];
145  uint8_t *dst;
146  int ch, out_size, ret;
147 
148  out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
149  if ((ret = ff_alloc_packet2(avctx, avpkt, out_size, 0)) < 0)
150  return ret;
151  dst = avpkt->data;
152 
153  if (!c->header_parsed) {
154  int hdrsize;
155  if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
156  av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
157  return AVERROR(EINVAL);
158  }
159  dst += hdrsize;
160  c->header_parsed = 1;
161  }
162 
163  for (ch = 0; ch < avctx->channels; ch++) {
164  adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
165  dst += BLOCK_SIZE;
166  }
167 
168  *got_packet_ptr = 1;
169  return 0;
170 }
171 
172  AVCodec ff_adpcm_adx_encoder = {
173  .name = "adpcm_adx",
174  .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
175  .type = AVMEDIA_TYPE_AUDIO,
176  .id = AV_CODEC_ID_ADPCM_ADX,
177  .priv_data_size = sizeof(ADXContext),
178  .init = adx_encode_init,
179  .encode2 = adx_encode_frame,
180  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
181  AV_SAMPLE_FMT_NONE },
182 };
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
put_sbits
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:240
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
channels
channels
Definition: aptx.c:30
AVPacket::size
int size
Definition: avcodec.h:1431
AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
ADXContext::coeff
int coeff[2]
Definition: adx.h:48
out_size
int out_size
Definition: movenc.c:55
BLOCK_SIZE
#define BLOCK_SIZE
Definition: adx.h:53
AVCodec
AVCodec.
Definition: avcodec.h:3408
adx_encode
static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav, ADXChannelState *prev, int channels)
Definition: adxenc.c:37
ff_adx_calculate_coeffs
void ff_adx_calculate_coeffs(int cutoff, int sample_rate, int bits, int *coeff)
Calculate LPC coefficients based on cutoff frequency and sample rate.
Definition: adx.c:26
ff_alloc_packet2
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
uint8_t
uint8_t
Definition: audio_convert.c:194
av_cold
#define av_cold
Definition: attributes.h:82
frame
static AVFrame * frame
Definition: demuxing_decoding.c:53
AVPacket::data
uint8_t * data
Definition: avcodec.h:1430
AV_WB16
#define AV_WB16(p, v)
Definition: intreadwrite.h:405
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ROUNDED_DIV
#define ROUNDED_DIV(a, b)
Definition: common.h:56
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
COEFF_BITS
#define COEFF_BITS
Definition: adx.h:51
s2
#define s2
Definition: regdef.h:39
HEADER_SIZE
#define HEADER_SIZE
Definition: adxenc.c:98
AVERROR
#define AVERROR(e)
Definition: error.h:43
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
s0
#define s0
Definition: regdef.h:37
AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3415
ADXContext::header_parsed
int header_parsed
Definition: adx.h:45
ff_adpcm_adx_encoder
AVCodec ff_adpcm_adx_encoder
Definition: adxenc.c:172
AV_CODEC_ID_ADPCM_ADX
Definition: avcodec.h:499
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2193
BLOCK_SAMPLES
#define BLOCK_SAMPLES
Definition: adx.h:54
avcodec.h
Libavcodec external API header.
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2173
AVCodecContext
main external API structure.
Definition: avcodec.h:1518
adx_encode_frame
static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: adxenc.c:140
buf
void * buf
Definition: avisynth_c.h:690
ADXContext
Definition: adx.h:42
ADXChannelState::s2
int s2
Definition: adx.h:39
s1
#define s1
Definition: regdef.h:38
ADXContext::cutoff
int cutoff
Definition: adx.h:47
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
internal.h
common internal api header.
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
if
if(ret< 0)
Definition: vf_mcdeint.c:279
AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
c
static double c[64]
Definition: vsrc_mptestsrc.c:87
ADXChannelState::s1
int s1
Definition: adx.h:39
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
adx_encode_init
static av_cold int adx_encode_init(AVCodecContext *avctx)
Definition: adxenc.c:123
bytestream_put_buffer
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:368
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1545
adx_encode_header
static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
Definition: adxenc.c:100
ADXContext::prev
ADXChannelState prev[2]
Definition: adx.h:44
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2174
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
adx.h
SEGA CRI adx codecs.
min
float min
Definition: vorbis_enc_data.h:456
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1407
ch
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
put_bits.h
bitstream writer API
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