FFmpeg  4.3
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 };
AVCodec
AVCodec.
Definition: codec.h:190
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
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
out_size
int out_size
Definition: movenc.c:55
ff_ws_snd1_decoder
AVCodec ff_ws_snd1_decoder
Definition: ws-snd1.c:173
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
data
const char data[16]
Definition: mxf.c:91
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
ws_snd_decode_init
static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
Definition: ws-snd1.c:44
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:90
intreadwrite.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AV_RL16
#define AV_RL16
Definition: intreadwrite.h:42
AV_SAMPLE_FMT_U8
AV_SAMPLE_FMT_U8
Definition: audio_convert.c:194
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
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
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
sample
#define sample
Definition: flacdsp_template.c:44
size
int size
Definition: twinvq_data.h:11134
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
code
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
Definition: filter_design.txt:178
common.h
uint8_t
uint8_t
Definition: audio_convert.c:194
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
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
AV_CODEC_ID_WESTWOOD_SND1
@ AV_CODEC_ID_WESTWOOD_SND1
Definition: codec_id.h:427
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
ws_snd_decode_frame
static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: ws-snd1.c:53
ws_adpcm_4bit
static const int8_t ws_adpcm_4bit[]
Definition: ws-snd1.c:39
AVPacket
This structure stores compressed data.
Definition: packet.h:332
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59