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 "codec_internal.h"
26 #include "encode.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 
85  d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
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->ch_layout.nb_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 
124  static av_cold int adx_encode_init(AVCodecContext *avctx)
125 {
126  ADXContext *c = avctx->priv_data;
127 
128  if (avctx->ch_layout.nb_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 channels = avctx->ch_layout.nb_channels;
148  int ch, out_size, ret;
149 
150  if (!samples) {
151  if (c->eof)
152  return 0;
153  if ((ret = ff_get_encode_buffer(avctx, avpkt, 18, 0)) < 0)
154  return ret;
155  c->eof = 1;
156  dst = avpkt->data;
157  bytestream_put_be16(&dst, 0x8001);
158  bytestream_put_be16(&dst, 0x000E);
159  bytestream_put_be64(&dst, 0x0);
160  bytestream_put_be32(&dst, 0x0);
161  bytestream_put_be16(&dst, 0x0);
162  *got_packet_ptr = 1;
163  return 0;
164  }
165 
166  out_size = BLOCK_SIZE * channels + !c->header_parsed * HEADER_SIZE;
167  if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
168  return ret;
169  dst = avpkt->data;
170 
171  if (!c->header_parsed) {
172  int hdrsize;
173  if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
174  av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
175  return AVERROR(EINVAL);
176  }
177  dst += hdrsize;
178  c->header_parsed = 1;
179  }
180 
181  for (ch = 0; ch < channels; ch++) {
182  adx_encode(c, dst, samples + ch, &c->prev[ch], channels);
183  dst += BLOCK_SIZE;
184  }
185 
186  *got_packet_ptr = 1;
187  return 0;
188 }
189 
190  const FFCodec ff_adpcm_adx_encoder = {
191  .p.name = "adpcm_adx",
192  CODEC_LONG_NAME("SEGA CRI ADX ADPCM"),
193  .p.type = AVMEDIA_TYPE_AUDIO,
194  .p.id = AV_CODEC_ID_ADPCM_ADX,
195  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
196  AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
197  .priv_data_size = sizeof(ADXContext),
198  .init = adx_encode_init,
199  FF_CODEC_ENCODE_CB(adx_encode_frame),
200  .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
201  AV_SAMPLE_FMT_NONE },
202  .caps_internal = FF_CODEC_CAP_EOF_FLUSH,
203 };
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1083
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:35
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1056
FF_CODEC_CAP_EOF_FLUSH
#define FF_CODEC_CAP_EOF_FLUSH
The encoder has AV_CODEC_CAP_DELAY set, but does not actually have delay - it only wants to be flushe...
Definition: codec_internal.h:90
put_sbits
static void put_sbits(PutBitContext *pb, int n, int32_t value)
Definition: put_bits.h:281
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:62
out_size
int out_size
Definition: movenc.c:56
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AVPacket::data
uint8_t * data
Definition: packet.h:539
encode.h
FFCodec
Definition: codec_internal.h:127
max
#define max(a, b)
Definition: cuda_runtime.h:33
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:321
ADXChannelState::s1
int s1
Definition: adx.h:35
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:131
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1071
FF_CODEC_ENCODE_CB
#define FF_CODEC_ENCODE_CB(func)
Definition: codec_internal.h:320
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
ff_adpcm_adx_encoder
const FFCodec ff_adpcm_adx_encoder
Definition: adxenc.c:190
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
BLOCK_SAMPLES
#define BLOCK_SAMPLES
Definition: adx.h:52
AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE
#define AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE
This encoder can reorder user opaque values from input AVFrames and return them with corresponding ou...
Definition: codec.h:159
channels
channels
Definition: aptx.h:31
PutBitContext
Definition: put_bits.h:50
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:296
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:121
AV_WB16
#define AV_WB16(p, v)
Definition: intreadwrite.h:401
ROUNDED_DIV
#define ROUNDED_DIV(a, b)
Definition: common.h:58
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:25
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts.c:368
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:376
AVPacket::size
int size
Definition: packet.h:540
codec_internal.h
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
HEADER_SIZE
#define HEADER_SIZE
Definition: adxenc.c:99
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:34
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:194
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:451
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:106
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:76
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:40
COEFF_BITS
#define COEFF_BITS
Definition: adx.h:49
adx.h
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:143
scale
static void scale(int *out, const int *in, const int w, const int h, const int shift)
Definition: intra.c:291
AVPacket
This structure stores compressed data.
Definition: packet.h:516
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:478
bytestream.h
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
BLOCK_SIZE
#define BLOCK_SIZE
Definition: adx.h:51
put_bits.h
min
float min
Definition: vorbis_enc_data.h:429
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