FFmpeg
libgsmdec.c
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1 /*
2  * Interface to libgsm for GSM decoding
3  * Copyright (c) 2005 Alban Bedel <albeu@free.fr>
4  * Copyright (c) 2006, 2007 Michel Bardiaux <mbardiaux@mediaxim.be>
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 /**
24  * @file
25  * Interface to libgsm for GSM decoding
26  */
27 
28 // The idiosyncrasies of GSM-in-WAV are explained at http://kbs.cs.tu-berlin.de/~jutta/toast.html
29 
30 #include "config.h"
31 #if HAVE_GSM_H
32 #include <gsm.h>
33 #else
34 #include <gsm/gsm.h>
35 #endif
36 
38 #include "libavutil/common.h"
39 
40 #include "avcodec.h"
41 #include "internal.h"
42 #include "gsm.h"
43 
44 typedef struct LibGSMDecodeContext {
45  struct gsm_state *state;
47 
50 
51  avctx->channels = 1;
53  if (!avctx->sample_rate)
54  avctx->sample_rate = 8000;
56 
57  s->state = gsm_create();
58 
59  switch(avctx->codec_id) {
60  case AV_CODEC_ID_GSM:
61  avctx->frame_size = GSM_FRAME_SIZE;
62  avctx->block_align = GSM_BLOCK_SIZE;
63  break;
64  case AV_CODEC_ID_GSM_MS: {
65  int one = 1;
66  gsm_option(s->state, GSM_OPT_WAV49, &one);
67  avctx->frame_size = 2 * GSM_FRAME_SIZE;
69  }
70  }
71 
72  return 0;
73 }
74 
77 
78  gsm_destroy(s->state);
79  s->state = NULL;
80  return 0;
81 }
82 
83 static int libgsm_decode_frame(AVCodecContext *avctx, void *data,
84  int *got_frame_ptr, AVPacket *avpkt)
85 {
86  int i, ret;
88  AVFrame *frame = data;
89  uint8_t *buf = avpkt->data;
90  int buf_size = avpkt->size;
91  int16_t *samples;
92 
93  if (buf_size < avctx->block_align) {
94  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
95  return AVERROR_INVALIDDATA;
96  }
97 
98  /* get output buffer */
99  frame->nb_samples = avctx->frame_size;
100  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
101  return ret;
102  samples = (int16_t *)frame->data[0];
103 
104  for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
105  if ((ret = gsm_decode(s->state, buf, samples)) < 0)
106  return -1;
107  buf += GSM_BLOCK_SIZE;
109  }
110 
111  *got_frame_ptr = 1;
112 
113  return avctx->block_align;
114 }
115 
116 static void libgsm_flush(AVCodecContext *avctx) {
117  LibGSMDecodeContext *s = avctx->priv_data;
118  int one = 1;
119 
120  gsm_destroy(s->state);
121  s->state = gsm_create();
122  if (avctx->codec_id == AV_CODEC_ID_GSM_MS)
123  gsm_option(s->state, GSM_OPT_WAV49, &one);
124 }
125 
126 #if CONFIG_LIBGSM_DECODER
128  .name = "libgsm",
129  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
130  .type = AVMEDIA_TYPE_AUDIO,
131  .id = AV_CODEC_ID_GSM,
132  .priv_data_size = sizeof(LibGSMDecodeContext),
134  .close = libgsm_decode_close,
136  .flush = libgsm_flush,
137  .capabilities = AV_CODEC_CAP_DR1,
138  .wrapper_name = "libgsm",
139 };
140 #endif
141 #if CONFIG_LIBGSM_MS_DECODER
143  .name = "libgsm_ms",
144  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
145  .type = AVMEDIA_TYPE_AUDIO,
146  .id = AV_CODEC_ID_GSM_MS,
147  .priv_data_size = sizeof(LibGSMDecodeContext),
149  .close = libgsm_decode_close,
151  .flush = libgsm_flush,
152  .capabilities = AV_CODEC_CAP_DR1,
153  .wrapper_name = "libgsm",
154 };
155 #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
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
LibGSMDecodeContext::state
struct gsm_state * state
Definition: libgsmdec.c:45
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
LibGSMDecodeContext
Definition: libgsmdec.c:44
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
GSM_FRAME_SIZE
#define GSM_FRAME_SIZE
Definition: gsm.h:30
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
ff_libgsm_ms_decoder
AVCodec ff_libgsm_ms_decoder
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
s
#define s(width, name)
Definition: cbs_vp9.c:257
gsm.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
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
NULL
#define NULL
Definition: coverity.c:32
AV_CODEC_ID_GSM
@ AV_CODEC_ID_GSM
as in Berlin toast format
Definition: codec_id.h:428
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
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:188
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
libgsm_flush
static void libgsm_flush(AVCodecContext *avctx)
Definition: libgsmdec.c:116
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
GSM_BLOCK_SIZE
#define GSM_BLOCK_SIZE
Definition: gsm.h:25
common.h
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
ff_libgsm_decoder
AVCodec ff_libgsm_decoder
AV_CODEC_ID_GSM_MS
@ AV_CODEC_ID_GSM_MS
Definition: codec_id.h:440
libgsm_decode_frame
static int libgsm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: libgsmdec.c:83
ret
ret
Definition: filter_design.txt:187
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
libgsm_decode_init
static av_cold int libgsm_decode_init(AVCodecContext *avctx)
Definition: libgsmdec.c:48
libgsm_decode_close
static av_cold int libgsm_decode_close(AVCodecContext *avctx)
Definition: libgsmdec.c:75
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
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