FFmpeg
libgsmdec.c
Go to the documentation of this file.
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 #include "config_components.h"
32 #if HAVE_GSM_H
33 #include <gsm.h>
34 #else
35 #include <gsm/gsm.h>
36 #endif
37 
39 #include "libavutil/common.h"
40 
41 #include "avcodec.h"
42 #include "codec_internal.h"
43 #include "decode.h"
44 #include "gsm.h"
45 
46 typedef struct LibGSMDecodeContext {
47  struct gsm_state *state;
49 
52 
55  if (!avctx->sample_rate)
56  avctx->sample_rate = 8000;
58 
59  s->state = gsm_create();
60 
61  switch(avctx->codec_id) {
62  case AV_CODEC_ID_GSM:
63  avctx->frame_size = GSM_FRAME_SIZE;
64  avctx->block_align = GSM_BLOCK_SIZE;
65  break;
66  case AV_CODEC_ID_GSM_MS: {
67  int one = 1;
68  gsm_option(s->state, GSM_OPT_WAV49, &one);
69  avctx->frame_size = 2 * GSM_FRAME_SIZE;
71  }
72  }
73 
74  return 0;
75 }
76 
79 
80  gsm_destroy(s->state);
81  s->state = NULL;
82  return 0;
83 }
84 
86  int *got_frame_ptr, AVPacket *avpkt)
87 {
88  int i, ret;
90  uint8_t *buf = avpkt->data;
91  int buf_size = avpkt->size;
92  int16_t *samples;
93 
94  if (buf_size < avctx->block_align) {
95  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
96  return AVERROR_INVALIDDATA;
97  }
98 
99  /* get output buffer */
100  frame->nb_samples = avctx->frame_size;
101  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
102  return ret;
103  samples = (int16_t *)frame->data[0];
104 
105  for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
106  if ((ret = gsm_decode(s->state, buf, samples)) < 0)
107  return -1;
108  buf += GSM_BLOCK_SIZE;
110  }
111 
112  *got_frame_ptr = 1;
113 
114  return avctx->block_align;
115 }
116 
117 static void libgsm_flush(AVCodecContext *avctx) {
118  LibGSMDecodeContext *s = avctx->priv_data;
119  int one = 1;
120 
121  gsm_destroy(s->state);
122  s->state = gsm_create();
123  if (avctx->codec_id == AV_CODEC_ID_GSM_MS)
124  gsm_option(s->state, GSM_OPT_WAV49, &one);
125 }
126 
127 #if CONFIG_LIBGSM_DECODER
128 const FFCodec ff_libgsm_decoder = {
129  .p.name = "libgsm",
130  CODEC_LONG_NAME("libgsm GSM"),
131  .p.type = AVMEDIA_TYPE_AUDIO,
132  .p.id = AV_CODEC_ID_GSM,
133  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
134  .p.wrapper_name = "libgsm",
135  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
136  .priv_data_size = sizeof(LibGSMDecodeContext),
138  .close = libgsm_decode_close,
140  .flush = libgsm_flush,
141 };
142 #endif
143 #if CONFIG_LIBGSM_MS_DECODER
145  .p.name = "libgsm_ms",
146  CODEC_LONG_NAME("libgsm GSM Microsoft variant"),
147  .p.type = AVMEDIA_TYPE_AUDIO,
148  .p.id = AV_CODEC_ID_GSM_MS,
149  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
150  .p.wrapper_name = "libgsm",
151  .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE,
152  .priv_data_size = sizeof(LibGSMDecodeContext),
154  .close = libgsm_decode_close,
156  .flush = libgsm_flush,
157 };
158 #endif
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1035
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1007
LibGSMDecodeContext::state
struct gsm_state * state
Definition: libgsmdec.c:47
LibGSMDecodeContext
Definition: libgsmdec.c:46
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
AVPacket::data
uint8_t * data
Definition: packet.h:374
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:356
FF_CODEC_CAP_NOT_INIT_THREADSAFE
#define FF_CODEC_CAP_NOT_INIT_THREADSAFE
The codec is not known to be init-threadsafe (i.e.
Definition: codec_internal.h:34
FFCodec
Definition: codec_internal.h:119
init
static int init
Definition: av_tx.c:47
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:123
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2059
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:180
av_cold
#define av_cold
Definition: attributes.h:90
ff_libgsm_decoder
const FFCodec ff_libgsm_decoder
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:298
s
#define s(width, name)
Definition: cbs_vp9.c:256
gsm.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
decode.h
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:264
GSM_MS_BLOCK_SIZE
#define GSM_MS_BLOCK_SIZE
Definition: gsm.h:26
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:408
NULL
#define NULL
Definition: coverity.c:32
ff_libgsm_ms_decoder
const FFCodec ff_libgsm_ms_decoder
AV_CODEC_ID_GSM
@ AV_CODEC_ID_GSM
as in Berlin toast format
Definition: codec_id.h:451
av_channel_layout_uninit
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Definition: channel_layout.c:630
AV_CODEC_CAP_CHANNEL_CONF
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:109
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1450
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:375
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:293
codec_internal.h
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1023
libgsm_flush
static void libgsm_flush(AVCodecContext *avctx)
Definition: libgsmdec.c:117
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
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
avcodec.h
AV_CODEC_ID_GSM_MS
@ AV_CODEC_ID_GSM_MS
Definition: codec_id.h:463
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:1052
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:50
libgsm_decode_close
static av_cold int libgsm_decode_close(AVCodecContext *avctx)
Definition: libgsmdec.c:77
AVCodecContext
main external API structure.
Definition: avcodec.h:398
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:351
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:425
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
libgsm_decode_frame
static int libgsm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: libgsmdec.c:85