FFmpeg  4.3
gsmdec.c
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1 /*
2  * gsm 06.10 decoder
3  * Copyright (c) 2010 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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  * GSM decoder
25  */
26 
28 #include "avcodec.h"
29 #include "get_bits.h"
30 #include "internal.h"
31 #include "msgsmdec.h"
32 
33 #include "gsmdec_template.c"
34 
35 static av_cold int gsm_init(AVCodecContext *avctx)
36 {
37  avctx->channels = 1;
39  if (!avctx->sample_rate)
40  avctx->sample_rate = 8000;
42 
43  switch (avctx->codec_id) {
44  case AV_CODEC_ID_GSM:
45  avctx->frame_size = GSM_FRAME_SIZE;
46  avctx->block_align = GSM_BLOCK_SIZE;
47  break;
48  case AV_CODEC_ID_GSM_MS:
49  avctx->frame_size = 2 * GSM_FRAME_SIZE;
50  if (!avctx->block_align)
52  else
53  if (avctx->block_align < MSN_MIN_BLOCK_SIZE ||
54  avctx->block_align > GSM_MS_BLOCK_SIZE ||
55  (avctx->block_align - MSN_MIN_BLOCK_SIZE) % 3) {
56  av_log(avctx, AV_LOG_ERROR, "Invalid block alignment %d\n",
57  avctx->block_align);
58  return AVERROR_INVALIDDATA;
59  }
60  }
61 
62  return 0;
63 }
64 
65 static int gsm_decode_frame(AVCodecContext *avctx, void *data,
66  int *got_frame_ptr, AVPacket *avpkt)
67 {
68  AVFrame *frame = data;
69  int res;
70  GetBitContext gb;
71  const uint8_t *buf = avpkt->data;
72  int buf_size = avpkt->size;
73  int16_t *samples;
74 
75  if (buf_size < avctx->block_align) {
76  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
77  return AVERROR_INVALIDDATA;
78  }
79 
80  /* get output buffer */
81  frame->nb_samples = avctx->frame_size;
82  if ((res = ff_get_buffer(avctx, frame, 0)) < 0)
83  return res;
84  samples = (int16_t *)frame->data[0];
85 
86  switch (avctx->codec_id) {
87  case AV_CODEC_ID_GSM:
88  init_get_bits(&gb, buf, buf_size * 8);
89  if (get_bits(&gb, 4) != 0xd)
90  av_log(avctx, AV_LOG_WARNING, "Missing GSM magic!\n");
91  res = gsm_decode_block(avctx, samples, &gb, GSM_13000);
92  if (res < 0)
93  return res;
94  break;
95  case AV_CODEC_ID_GSM_MS:
96  res = ff_msgsm_decode_block(avctx, samples, buf,
97  (GSM_MS_BLOCK_SIZE - avctx->block_align) / 3);
98  if (res < 0)
99  return res;
100  }
101 
102  *got_frame_ptr = 1;
103 
104  return avctx->block_align;
105 }
106 
107 static void gsm_flush(AVCodecContext *avctx)
108 {
109  GSMContext *s = avctx->priv_data;
110  memset(s, 0, sizeof(*s));
111 }
112 
113 #if CONFIG_GSM_DECODER
115  .name = "gsm",
116  .long_name = NULL_IF_CONFIG_SMALL("GSM"),
117  .type = AVMEDIA_TYPE_AUDIO,
118  .id = AV_CODEC_ID_GSM,
119  .priv_data_size = sizeof(GSMContext),
120  .init = gsm_init,
122  .flush = gsm_flush,
123  .capabilities = AV_CODEC_CAP_DR1,
124 };
125 #endif
126 #if CONFIG_GSM_MS_DECODER
128  .name = "gsm_ms",
129  .long_name = NULL_IF_CONFIG_SMALL("GSM Microsoft variant"),
130  .type = AVMEDIA_TYPE_AUDIO,
131  .id = AV_CODEC_ID_GSM_MS,
132  .priv_data_size = sizeof(GSMContext),
133  .init = gsm_init,
135  .flush = gsm_flush,
136  .capabilities = AV_CODEC_CAP_DR1,
137 };
138 #endif
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1206
AVCodec
AVCodec.
Definition: codec.h:190
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
gsm_init
static av_cold int gsm_init(AVCodecContext *avctx)
Definition: gsmdec.c:35
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
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1186
gsmdec_template.c
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
GSM_13000
@ GSM_13000
Definition: gsm.h:33
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
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
ff_msgsm_decode_block
int ff_msgsm_decode_block(AVCodecContext *avctx, int16_t *samples, const uint8_t *buf, int mode)
Definition: msgsmdec.c:29
GetBitContext
Definition: get_bits.h:61
GSM_FRAME_SIZE
#define GSM_FRAME_SIZE
Definition: gsm.h:30
GSM_BLOCK_SIZE
#define GSM_BLOCK_SIZE
Definition: gsmdec.c:28
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
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
ff_gsm_decoder
AVCodec ff_gsm_decoder
s
#define s(width, name)
Definition: cbs_vp9.c:257
msgsmdec.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
gsm_decode_frame
static int gsm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: gsmdec.c:65
get_bits.h
GSM_MS_BLOCK_SIZE
#define GSM_MS_BLOCK_SIZE
Definition: gsm.h:26
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:536
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:500
AV_CODEC_ID_GSM
@ AV_CODEC_ID_GSM
as in Berlin toast format
Definition: codec_id.h:428
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
MSN_MIN_BLOCK_SIZE
#define MSN_MIN_BLOCK_SIZE
Definition: gsm.h:27
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
gsm_decode_block
static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples, GetBitContext *gb, int mode)
Definition: gsmdec_template.c:122
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
AV_CODEC_ID_GSM_MS
@ AV_CODEC_ID_GSM_MS
Definition: codec_id.h:440
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1223
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
gsm_flush
static void gsm_flush(AVCodecContext *avctx)
Definition: gsmdec.c:107
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
ff_gsm_ms_decoder
AVCodec ff_gsm_ms_decoder
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
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
GSMContext
Definition: gsmdec_data.h:28