FFmpeg  4.3
dsicinaudio.c
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1 /*
2  * Delphine Software International CIN audio decoder
3  * Copyright (c) 2006 Gregory Montoir (cyx@users.sourceforge.net)
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 /**
23  * @file
24  * Delphine Software International CIN audio decoder
25  */
26 
28 
29 #include "avcodec.h"
30 #include "bytestream.h"
31 #include "internal.h"
32 #include "mathops.h"
33 
34 typedef struct CinAudioContext {
36  int delta;
38 
39 
40 /* table defining a geometric sequence with multiplier = 32767 ^ (1 / 128) */
41 static const int16_t cinaudio_delta16_table[256] = {
42  0, 0, 0, 0, 0, 0, 0, 0,
43  0, 0, 0, 0, 0, 0, 0, 0,
44  0, 0, 0, -30210, -27853, -25680, -23677, -21829,
45  -20126, -18556, -17108, -15774, -14543, -13408, -12362, -11398,
46  -10508, -9689, -8933, -8236, -7593, -7001, -6455, -5951,
47  -5487, -5059, -4664, -4300, -3964, -3655, -3370, -3107,
48  -2865, -2641, -2435, -2245, -2070, -1908, -1759, -1622,
49  -1495, -1379, -1271, -1172, -1080, -996, -918, -847,
50  -781, -720, -663, -612, -564, -520, -479, -442,
51  -407, -376, -346, -319, -294, -271, -250, -230,
52  -212, -196, -181, -166, -153, -141, -130, -120,
53  -111, -102, -94, -87, -80, -74, -68, -62,
54  -58, -53, -49, -45, -41, -38, -35, -32,
55  -30, -27, -25, -23, -21, -20, -18, -17,
56  -15, -14, -13, -12, -11, -10, -9, -8,
57  -7, -6, -5, -4, -3, -2, -1, 0,
58  0, 1, 2, 3, 4, 5, 6, 7,
59  8, 9, 10, 11, 12, 13, 14, 15,
60  17, 18, 20, 21, 23, 25, 27, 30,
61  32, 35, 38, 41, 45, 49, 53, 58,
62  62, 68, 74, 80, 87, 94, 102, 111,
63  120, 130, 141, 153, 166, 181, 196, 212,
64  230, 250, 271, 294, 319, 346, 376, 407,
65  442, 479, 520, 564, 612, 663, 720, 781,
66  847, 918, 996, 1080, 1172, 1271, 1379, 1495,
67  1622, 1759, 1908, 2070, 2245, 2435, 2641, 2865,
68  3107, 3370, 3655, 3964, 4300, 4664, 5059, 5487,
69  5951, 6455, 7001, 7593, 8236, 8933, 9689, 10508,
70  11398, 12362, 13408, 14543, 15774, 17108, 18556, 20126,
71  21829, 23677, 25680, 27853, 30210, 0, 0, 0,
72  0, 0, 0, 0, 0, 0, 0, 0,
73  0, 0, 0, 0, 0, 0, 0, 0
74 };
75 
77 {
78  CinAudioContext *cin = avctx->priv_data;
79 
80  cin->initial_decode_frame = 1;
81  cin->delta = 0;
83  avctx->channels = 1;
85 
86  return 0;
87 }
88 
89 static int cinaudio_decode_frame(AVCodecContext *avctx, void *data,
90  int *got_frame_ptr, AVPacket *avpkt)
91 {
92  AVFrame *frame = data;
93  const uint8_t *buf = avpkt->data;
94  CinAudioContext *cin = avctx->priv_data;
95  const uint8_t *buf_end = buf + avpkt->size;
96  int16_t *samples;
97  int delta, ret;
98 
99  /* get output buffer */
100  frame->nb_samples = avpkt->size - cin->initial_decode_frame;
101  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
102  return ret;
103  samples = (int16_t *)frame->data[0];
104 
105  delta = cin->delta;
106  if (cin->initial_decode_frame) {
107  cin->initial_decode_frame = 0;
108  delta = sign_extend(AV_RL16(buf), 16);
109  buf += 2;
110  *samples++ = delta;
111  }
112  while (buf < buf_end) {
113  delta += cinaudio_delta16_table[*buf++];
114  delta = av_clip_int16(delta);
115  *samples++ = delta;
116  }
117  cin->delta = delta;
118 
119  *got_frame_ptr = 1;
120 
121  return avpkt->size;
122 }
123 
125  .name = "dsicinaudio",
126  .long_name = NULL_IF_CONFIG_SMALL("Delphine Software International CIN audio"),
127  .type = AVMEDIA_TYPE_AUDIO,
129  .priv_data_size = sizeof(CinAudioContext),
132  .capabilities = AV_CODEC_CAP_DR1,
133 };
AVCodec
AVCodec.
Definition: codec.h:190
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
CinAudioContext::delta
int delta
Definition: dsicinaudio.c:36
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
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
CinAudioContext
Definition: dsicinaudio.c:34
av_cold
#define av_cold
Definition: attributes.h:90
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
AV_CODEC_ID_DSICINAUDIO
@ AV_CODEC_ID_DSICINAUDIO
Definition: codec_id.h:436
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AV_RL16
#define AV_RL16
Definition: intreadwrite.h:42
if
if(ret)
Definition: filter_design.txt:179
mathops.h
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
cinaudio_decode_frame
static int cinaudio_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: dsicinaudio.c:89
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
cinaudio_decode_init
static av_cold int cinaudio_decode_init(AVCodecContext *avctx)
Definition: dsicinaudio.c:76
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
delta
float delta
Definition: vorbis_enc_data.h:457
uint8_t
uint8_t
Definition: audio_convert.c:194
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: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
ff_dsicinaudio_decoder
AVCodec ff_dsicinaudio_decoder
Definition: dsicinaudio.c:124
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
cinaudio_delta16_table
static const int16_t cinaudio_delta16_table[256]
Definition: dsicinaudio.c:41
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
bytestream.h
CinAudioContext::initial_decode_frame
int initial_decode_frame
Definition: dsicinaudio.c:35