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 "encode.h"
26 #include "internal.h"
27 #include "put_bits.h"
28 
29 /**
30  * @file
31  * SEGA CRI adx codecs.
32  *
33  * Reference documents:
34  * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
35  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
36  */
37 
38 static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
39  ADXChannelState *prev, int channels)
40 {
41  PutBitContext pb;
42  int scale;
43  int i, j;
44  int s0, s1, s2, d;
45  int max = 0;
46  int min = 0;
47 
48  s1 = prev->s1;
49  s2 = prev->s2;
50  for (i = 0, j = 0; j < 32; i += channels, j++) {
51  s0 = wav[i];
52  d = s0 + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
53  if (max < d)
54  max = d;
55  if (min > d)
56  min = d;
57  s2 = s1;
58  s1 = s0;
59  }
60 
61  if (max == 0 && min == 0) {
62  prev->s1 = s1;
63  prev->s2 = s2;
64  memset(adx, 0, BLOCK_SIZE);
65  return;
66  }
67 
68  if (max / 7 > -min / 8)
69  scale = max / 7;
70  else
71  scale = -min / 8;
72 
73  if (scale == 0)
74  scale = 1;
75 
76  AV_WB16(adx, scale);
77 
78  init_put_bits(&pb, adx + 2, 16);
79 
80  s1 = prev->s1;
81  s2 = prev->s2;
82  for (i = 0, j = 0; j < 32; i += channels, j++) {
83  d = wav[i] + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
84 
86 
87  put_sbits(&pb, 4, d);
88 
89  s0 = d * scale + ((c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS);
90  s2 = s1;
91  s1 = s0;
92  }
93  prev->s1 = s1;
94  prev->s2 = s2;
95 
96  flush_put_bits(&pb);
97 }
98 
99 #define HEADER_SIZE 36
100 
101 static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
102 {
103  ADXContext *c = avctx->priv_data;
104 
105  bytestream_put_be16(&buf, 0x8000); /* header signature */
106  bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
107  bytestream_put_byte(&buf, 3); /* encoding */
108  bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
109  bytestream_put_byte(&buf, 4); /* sample size */
110  bytestream_put_byte(&buf, avctx->channels); /* channels */
111  bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
112  bytestream_put_be32(&buf, 0); /* total sample count */
113  bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
114  bytestream_put_byte(&buf, 3); /* version */
115  bytestream_put_byte(&buf, 0); /* flags */
116  bytestream_put_be32(&buf, 0); /* unknown */
117  bytestream_put_be32(&buf, 0); /* loop enabled */
118  bytestream_put_be16(&buf, 0); /* padding */
119  bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
120 
121  return HEADER_SIZE;
122 }
123 
125 {
126  ADXContext *c = avctx->priv_data;
127 
128  if (avctx->channels > 2) {
129  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
130  return AVERROR(EINVAL);
131  }
132  avctx->frame_size = BLOCK_SAMPLES;
133 
134  /* the cutoff can be adjusted, but this seems to work pretty well */
135  c->cutoff = 500;
136  ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
137 
138  return 0;
139 }
140 
141 static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
142  const AVFrame *frame, int *got_packet_ptr)
143 {
144  ADXContext *c = avctx->priv_data;
145  const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
146  uint8_t *dst;
147  int ch, out_size, ret;
148 
149  if (!samples) {
150  if (c->eof)
151  return 0;
152  if ((ret = ff_get_encode_buffer(avctx, avpkt, 18, 0)) < 0)
153  return ret;
154  c->eof = 1;
155  dst = avpkt->data;
156  bytestream_put_be16(&dst, 0x8001);
157  bytestream_put_be16(&dst, 0x000E);
158  bytestream_put_be64(&dst, 0x0);
159  bytestream_put_be32(&dst, 0x0);
160  bytestream_put_be16(&dst, 0x0);
161  *got_packet_ptr = 1;
162  return 0;
163  }
164 
165  out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
166  if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
167  return ret;
168  dst = avpkt->data;
169 
170  if (!c->header_parsed) {
171  int hdrsize;
172  if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
173  av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
174  return AVERROR(EINVAL);
175  }
176  dst += hdrsize;
177  c->header_parsed = 1;
178  }
179 
180  for (ch = 0; ch < avctx->channels; ch++) {
181  adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
182  dst += BLOCK_SIZE;
183  }
184 
185  avpkt->pts = frame->pts;
186  avpkt->duration = frame->nb_samples;
187  *got_packet_ptr = 1;
188  return 0;
189 }
190 
192  .name = "adpcm_adx",
193  .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
194  .type = AVMEDIA_TYPE_AUDIO,
195  .id = AV_CODEC_ID_ADPCM_ADX,
196  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
197  .priv_data_size = sizeof(ADXContext),
199  .encode2 = adx_encode_frame,
200  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
202  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
203 };
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1012
AVCodec
AVCodec.
Definition: codec.h:202
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:42
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
ADXChannelState::s2
int s2
Definition: adx.h:39
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:992
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
put_sbits
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:280
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:61
out_size
int out_size
Definition: movenc.c:55
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:310
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
encode.h
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:391
max
#define max(a, b)
Definition: cuda_runtime.h:33
ADXChannelState::s1
int s1
Definition: adx.h:39
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1388
adx_encode_frame
static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: adxenc.c:141
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
adx_encode
static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav, ADXChannelState *prev, int channels)
Definition: adxenc.c:38
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
s1
#define s1
Definition: regdef.h:38
BLOCK_SAMPLES
#define BLOCK_SAMPLES
Definition: adx.h:54
channels
channels
Definition: aptx.h:33
PutBitContext
Definition: put_bits.h:49
if
if(ret)
Definition: filter_design.txt:179
NULL
#define NULL
Definition: coverity.c:32
av_clip_intp2
#define av_clip_intp2
Definition: common.h:117
AV_WB16
#define AV_WB16(p, v)
Definition: intreadwrite.h:405
ROUNDED_DIV
#define ROUNDED_DIV(a, b)
Definition: common.h:49
adx_encode_init
static av_cold int adx_encode_init(AVCodecContext *avctx)
Definition: adxenc.c:124
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
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
s2
#define s2
Definition: regdef.h:39
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AV_CODEC_ID_ADPCM_ADX
@ AV_CODEC_ID_ADPCM_ADX
Definition: codec_id.h:362
AVPacket::size
int size
Definition: packet.h:374
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:117
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
HEADER_SIZE
#define HEADER_SIZE
Definition: adxenc.c:99
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:993
bytestream_put_buffer
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:372
ADXChannelState
Definition: adx.h:38
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:366
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:209
adx_encode_header
static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
Definition: adxenc.c:101
avcodec.h
ret
ret
Definition: filter_design.txt:187
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
AVCodecContext
main external API structure.
Definition: avcodec.h:383
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:78
ff_adpcm_adx_encoder
const AVCodec ff_adpcm_adx_encoder
Definition: adxenc.c:191
AV_CODEC_CAP_DELAY
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: codec.h:82
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
ADXContext
Definition: adx.h:42
COEFF_BITS
#define COEFF_BITS
Definition: adx.h:51
adx.h
s0
#define s0
Definition: regdef.h:37
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:142
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
d
d
Definition: ffmpeg_filter.c:156
bytestream.h
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
BLOCK_SIZE
#define BLOCK_SIZE
Definition: adx.h:53
put_bits.h
min
float min
Definition: vorbis_enc_data.h:429