FFmpeg
ws-snd1.c
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1 /*
2  * Westwood SNDx codecs
3  * Copyright (c) 2005 Konstantin Shishkov
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 <stdint.h>
23 
25 #include "libavutil/common.h"
26 #include "libavutil/intreadwrite.h"
27 #include "avcodec.h"
28 #include "internal.h"
29 
30 /**
31  * @file
32  * Westwood SNDx codecs
33  *
34  * Reference documents about VQA format and its audio codecs
35  * can be found here:
36  * http://www.multimedia.cx
37  */
38 
39 static const int8_t ws_adpcm_4bit[] = {
40  -9, -8, -6, -5, -4, -3, -2, -1,
41  0, 1, 2, 3, 4, 5, 6, 8
42 };
43 
45 {
46  avctx->channels = 1;
49 
50  return 0;
51 }
52 
53 static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
54  int *got_frame_ptr, AVPacket *avpkt)
55 {
56  AVFrame *frame = data;
57  const uint8_t *buf = avpkt->data;
58  int buf_size = avpkt->size;
59 
60  int in_size, out_size, ret;
61  int sample = 128;
63  uint8_t *samples_end;
64 
65  if (!buf_size)
66  return 0;
67 
68  if (buf_size < 4) {
69  av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
70  return AVERROR(EINVAL);
71  }
72 
73  out_size = AV_RL16(&buf[0]);
74  in_size = AV_RL16(&buf[2]);
75  buf += 4;
76 
77  if (in_size > buf_size) {
78  av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n");
79  return AVERROR_INVALIDDATA;
80  }
81 
82  /* get output buffer */
83  frame->nb_samples = out_size;
84  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
85  return ret;
86  samples = frame->data[0];
87  samples_end = samples + out_size;
88 
89  if (in_size == out_size) {
90  memcpy(samples, buf, out_size);
91  *got_frame_ptr = 1;
92  return buf_size;
93  }
94 
95  while (samples < samples_end && buf - avpkt->data < buf_size) {
96  int code, smp, size;
97  uint8_t count;
98  code = *buf >> 6;
99  count = *buf & 0x3F;
100  buf++;
101 
102  /* make sure we don't write past the output buffer */
103  switch (code) {
104  case 0: smp = 4 * (count + 1); break;
105  case 1: smp = 2 * (count + 1); break;
106  case 2: smp = (count & 0x20) ? 1 : count + 1; break;
107  default: smp = count + 1; break;
108  }
109  if (samples_end - samples < smp)
110  break;
111 
112  /* make sure we don't read past the input buffer */
113  size = ((code == 2 && (count & 0x20)) || code == 3) ? 0 : count + 1;
114  if ((buf - avpkt->data) + size > buf_size)
115  break;
116 
117  switch (code) {
118  case 0: /* ADPCM 2-bit */
119  for (count++; count > 0; count--) {
120  code = *buf++;
121  sample += ( code & 0x3) - 2;
122  sample = av_clip_uint8(sample);
123  *samples++ = sample;
124  sample += ((code >> 2) & 0x3) - 2;
125  sample = av_clip_uint8(sample);
126  *samples++ = sample;
127  sample += ((code >> 4) & 0x3) - 2;
128  sample = av_clip_uint8(sample);
129  *samples++ = sample;
130  sample += (code >> 6) - 2;
131  sample = av_clip_uint8(sample);
132  *samples++ = sample;
133  }
134  break;
135  case 1: /* ADPCM 4-bit */
136  for (count++; count > 0; count--) {
137  code = *buf++;
138  sample += ws_adpcm_4bit[code & 0xF];
139  sample = av_clip_uint8(sample);
140  *samples++ = sample;
141  sample += ws_adpcm_4bit[code >> 4];
142  sample = av_clip_uint8(sample);
143  *samples++ = sample;
144  }
145  break;
146  case 2: /* no compression */
147  if (count & 0x20) { /* big delta */
148  int8_t t;
149  t = count;
150  t <<= 3;
151  sample += t >> 3;
152  sample = av_clip_uint8(sample);
153  *samples++ = sample;
154  } else { /* copy */
155  memcpy(samples, buf, smp);
156  samples += smp;
157  buf += smp;
158  sample = buf[-1];
159  }
160  break;
161  default: /* run */
162  memset(samples, sample, smp);
163  samples += smp;
164  }
165  }
166 
167  frame->nb_samples = samples - frame->data[0];
168  *got_frame_ptr = 1;
169 
170  return buf_size;
171 }
172 
174  .name = "ws_snd1",
175  .long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"),
176  .type = AVMEDIA_TYPE_AUDIO,
178  .init = ws_snd_decode_init,
179  .decode = ws_snd_decode_frame,
180  .capabilities = AV_CODEC_CAP_DR1,
181 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
int size
Definition: avcodec.h:1481
int out_size
Definition: movenc.c:55
static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
Definition: ws-snd1.c:44
#define sample
AVCodec.
Definition: avcodec.h:3492
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
Definition: bytestream.h:87
AVCodec ff_ws_snd1_decoder
Definition: ws-snd1.c:173
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
AV_SAMPLE_FMT_U8
uint8_t * data
Definition: avcodec.h:1480
ptrdiff_t size
Definition: opengl_enc.c:100
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * name
Name of the codec implementation.
Definition: avcodec.h:3499
GLsizei count
Definition: opengl_enc.c:108
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2279
audio channel layout utility functions
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
Libavcodec external API header.
main external API structure.
Definition: avcodec.h:1568
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
void * buf
Definition: avisynth_c.h:766
static const int8_t ws_adpcm_4bit[]
Definition: ws-snd1.c:39
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
common internal api header.
common internal and external API header
int channels
number of audio channels
Definition: avcodec.h:2229
Filter the word “frame” indicates either a video frame or a group of audio samples
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
#define AV_CH_LAYOUT_MONO
static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: ws-snd1.c:53
This structure stores compressed data.
Definition: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:984