FFmpeg
libgsmenc.c
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1 /*
2  * Interface to libgsm for GSM encoding
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 encoding
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 "encode.h"
42 #include "internal.h"
43 #include "gsm.h"
44 
46  gsm_destroy(avctx->priv_data);
47  avctx->priv_data = NULL;
48  return 0;
49 }
50 
52  if (avctx->channels > 1) {
53  av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
54  avctx->channels);
55  return -1;
56  }
57 
58  if (avctx->sample_rate != 8000) {
59  av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
60  avctx->sample_rate);
62  return -1;
63  }
64  if (avctx->bit_rate != 13000 /* Official */ &&
65  avctx->bit_rate != 13200 /* Very common */ &&
66  avctx->bit_rate != 0 /* Unknown; a.o. mov does not set bitrate when decoding */ ) {
67  av_log(avctx, AV_LOG_ERROR, "Bitrate 13000bps required for GSM, got %"PRId64"bps\n",
68  avctx->bit_rate);
70  return -1;
71  }
72 
73  avctx->priv_data = gsm_create();
74  if (!avctx->priv_data)
75  goto error;
76 
77  switch(avctx->codec_id) {
78  case AV_CODEC_ID_GSM:
79  avctx->frame_size = GSM_FRAME_SIZE;
80  avctx->block_align = GSM_BLOCK_SIZE;
81  break;
82  case AV_CODEC_ID_GSM_MS: {
83  int one = 1;
84  gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
85  avctx->frame_size = 2*GSM_FRAME_SIZE;
87  }
88  }
89 
90  return 0;
91 error:
92  libgsm_encode_close(avctx);
93  return -1;
94 }
95 
96 static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
97  const AVFrame *frame, int *got_packet_ptr)
98 {
99  int ret;
100  gsm_signal *samples = (gsm_signal *)frame->data[0];
101  struct gsm_state *state = avctx->priv_data;
102 
103  if ((ret = ff_get_encode_buffer(avctx, avpkt, avctx->block_align, 0)) < 0)
104  return ret;
105 
106  switch(avctx->codec_id) {
107  case AV_CODEC_ID_GSM:
108  gsm_encode(state, samples, avpkt->data);
109  break;
110  case AV_CODEC_ID_GSM_MS:
111  gsm_encode(state, samples, avpkt->data);
112  gsm_encode(state, samples + GSM_FRAME_SIZE, avpkt->data + 32);
113  }
114 
115  *got_packet_ptr = 1;
116  return 0;
117 }
118 
119 static const AVCodecDefault libgsm_defaults[] = {
120  { "b", "13000" },
121  { NULL },
122 };
123 
124 #if CONFIG_LIBGSM_ENCODER
125 const AVCodec ff_libgsm_encoder = {
126  .name = "libgsm",
127  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
128  .type = AVMEDIA_TYPE_AUDIO,
129  .id = AV_CODEC_ID_GSM,
130  .capabilities = AV_CODEC_CAP_DR1,
131  .init = libgsm_encode_init,
132  .encode2 = libgsm_encode_frame,
133  .close = libgsm_encode_close,
134  .defaults = libgsm_defaults,
135  .channel_layouts= (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 },
136  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
138  .wrapper_name = "libgsm",
139 };
140 #endif
141 #if CONFIG_LIBGSM_MS_ENCODER
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  .capabilities = AV_CODEC_CAP_DR1,
148  .init = libgsm_encode_init,
149  .encode2 = libgsm_encode_frame,
150  .close = libgsm_encode_close,
151  .defaults = libgsm_defaults,
152  .channel_layouts= (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 },
153  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
155  .wrapper_name = "libgsm",
156 };
157 #endif
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:31
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1012
AVCodec
AVCodec.
Definition: codec.h:202
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:992
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:90
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:310
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
encode.h
FF_COMPLIANCE_UNOFFICIAL
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
Definition: avcodec.h:1284
state
static struct @319 state
GSM_FRAME_SIZE
#define GSM_FRAME_SIZE
Definition: gsm.h:30
libgsm_defaults
static const AVCodecDefault libgsm_defaults[]
Definition: libgsmenc.c:119
libgsm_encode_init
static av_cold int libgsm_encode_init(AVCodecContext *avctx)
Definition: libgsmenc.c:51
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
gsm.h
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
libgsm_encode_frame
static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libgsmenc.c:96
GSM_MS_BLOCK_SIZE
#define GSM_MS_BLOCK_SIZE
Definition: gsm.h:26
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:393
if
if(ret)
Definition: filter_design.txt:179
AVCodecDefault
Definition: internal.h:215
NULL
#define NULL
Definition: coverity.c:32
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:433
ff_libgsm_encoder
const AVCodec ff_libgsm_encoder
AV_CODEC_ID_GSM
@ AV_CODEC_ID_GSM
as in Berlin toast format
Definition: codec_id.h:441
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
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:117
ff_libgsm_ms_encoder
const AVCodec ff_libgsm_ms_encoder
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:993
GSM_BLOCK_SIZE
#define GSM_BLOCK_SIZE
Definition: gsm.h:25
common.h
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
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:209
libgsm_encode_close
static av_cold int libgsm_encode_close(AVCodecContext *avctx)
Definition: libgsmenc.c:45
avcodec.h
AV_CODEC_ID_GSM_MS
@ AV_CODEC_ID_GSM_MS
Definition: codec_id.h:453
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:1029
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
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1280
AVCodecContext
main external API structure.
Definition: avcodec.h:383
channel_layout.h
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:78
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:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28