libgsm.c
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1 /*
2  * Interface to libgsm for gsm encoding/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 encoding/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 #include "avcodec.h"
40 #include "internal.h"
41 #include "gsm.h"
42 
44  gsm_destroy(avctx->priv_data);
45  avctx->priv_data = NULL;
46  return 0;
47 }
48 
50  if (avctx->channels > 1) {
51  av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
52  avctx->channels);
53  return -1;
54  }
55 
56  if (avctx->sample_rate != 8000) {
57  av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
58  avctx->sample_rate);
60  return -1;
61  }
62  if (avctx->bit_rate != 13000 /* Official */ &&
63  avctx->bit_rate != 13200 /* Very common */ &&
64  avctx->bit_rate != 0 /* Unknown; a.o. mov does not set bitrate when decoding */ ) {
65  av_log(avctx, AV_LOG_ERROR, "Bitrate 13000bps required for GSM, got %dbps\n",
66  avctx->bit_rate);
68  return -1;
69  }
70 
71  avctx->priv_data = gsm_create();
72  if (!avctx->priv_data)
73  goto error;
74 
75  switch(avctx->codec_id) {
76  case AV_CODEC_ID_GSM:
77  avctx->frame_size = GSM_FRAME_SIZE;
78  avctx->block_align = GSM_BLOCK_SIZE;
79  break;
80  case AV_CODEC_ID_GSM_MS: {
81  int one = 1;
82  gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
83  avctx->frame_size = 2*GSM_FRAME_SIZE;
85  }
86  }
87 
88  return 0;
89 error:
90  libgsm_encode_close(avctx);
91  return -1;
92 }
93 
94 static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
95  const AVFrame *frame, int *got_packet_ptr)
96 {
97  int ret;
98  gsm_signal *samples = (gsm_signal *)frame->data[0];
99  struct gsm_state *state = avctx->priv_data;
100 
101  if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)) < 0)
102  return ret;
103 
104  switch(avctx->codec_id) {
105  case AV_CODEC_ID_GSM:
106  gsm_encode(state, samples, avpkt->data);
107  break;
108  case AV_CODEC_ID_GSM_MS:
109  gsm_encode(state, samples, avpkt->data);
110  gsm_encode(state, samples + GSM_FRAME_SIZE, avpkt->data + 32);
111  }
112 
113  *got_packet_ptr = 1;
114  return 0;
115 }
116 
117 
118 #if CONFIG_LIBGSM_ENCODER
119 AVCodec ff_libgsm_encoder = {
120  .name = "libgsm",
121  .type = AVMEDIA_TYPE_AUDIO,
122  .id = AV_CODEC_ID_GSM,
123  .init = libgsm_encode_init,
124  .encode2 = libgsm_encode_frame,
125  .close = libgsm_encode_close,
126  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
128  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
129 };
130 #endif
131 #if CONFIG_LIBGSM_MS_ENCODER
132 AVCodec ff_libgsm_ms_encoder = {
133  .name = "libgsm_ms",
134  .type = AVMEDIA_TYPE_AUDIO,
135  .id = AV_CODEC_ID_GSM_MS,
136  .init = libgsm_encode_init,
137  .encode2 = libgsm_encode_frame,
138  .close = libgsm_encode_close,
139  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
141  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
142 };
143 #endif
144 
145 typedef struct LibGSMDecodeContext {
146  struct gsm_state *state;
148 
150  LibGSMDecodeContext *s = avctx->priv_data;
151 
152  avctx->channels = 1;
154  if (!avctx->sample_rate)
155  avctx->sample_rate = 8000;
156  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
157 
158  s->state = gsm_create();
159 
160  switch(avctx->codec_id) {
161  case AV_CODEC_ID_GSM:
162  avctx->frame_size = GSM_FRAME_SIZE;
163  avctx->block_align = GSM_BLOCK_SIZE;
164  break;
165  case AV_CODEC_ID_GSM_MS: {
166  int one = 1;
167  gsm_option(s->state, GSM_OPT_WAV49, &one);
168  avctx->frame_size = 2 * GSM_FRAME_SIZE;
170  }
171  }
172 
173  return 0;
174 }
175 
177  LibGSMDecodeContext *s = avctx->priv_data;
178 
179  gsm_destroy(s->state);
180  s->state = NULL;
181  return 0;
182 }
183 
184 static int libgsm_decode_frame(AVCodecContext *avctx, void *data,
185  int *got_frame_ptr, AVPacket *avpkt)
186 {
187  int i, ret;
188  LibGSMDecodeContext *s = avctx->priv_data;
189  AVFrame *frame = data;
190  uint8_t *buf = avpkt->data;
191  int buf_size = avpkt->size;
192  int16_t *samples;
193 
194  if (buf_size < avctx->block_align) {
195  av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
196  return AVERROR_INVALIDDATA;
197  }
198 
199  /* get output buffer */
200  frame->nb_samples = avctx->frame_size;
201  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
202  return ret;
203  samples = (int16_t *)frame->data[0];
204 
205  for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
206  if ((ret = gsm_decode(s->state, buf, samples)) < 0)
207  return -1;
208  buf += GSM_BLOCK_SIZE;
209  samples += GSM_FRAME_SIZE;
210  }
211 
212  *got_frame_ptr = 1;
213 
214  return avctx->block_align;
215 }
216 
217 static void libgsm_flush(AVCodecContext *avctx) {
218  LibGSMDecodeContext *s = avctx->priv_data;
219  int one = 1;
220 
221  gsm_destroy(s->state);
222  s->state = gsm_create();
223  if (avctx->codec_id == AV_CODEC_ID_GSM_MS)
224  gsm_option(s->state, GSM_OPT_WAV49, &one);
225 }
226 
227 #if CONFIG_LIBGSM_DECODER
228 AVCodec ff_libgsm_decoder = {
229  .name = "libgsm",
230  .type = AVMEDIA_TYPE_AUDIO,
231  .id = AV_CODEC_ID_GSM,
232  .priv_data_size = sizeof(LibGSMDecodeContext),
236  .flush = libgsm_flush,
237  .capabilities = CODEC_CAP_DR1,
238  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
239 };
240 #endif
241 #if CONFIG_LIBGSM_MS_DECODER
242 AVCodec ff_libgsm_ms_decoder = {
243  .name = "libgsm_ms",
244  .type = AVMEDIA_TYPE_AUDIO,
245  .id = AV_CODEC_ID_GSM_MS,
246  .priv_data_size = sizeof(LibGSMDecodeContext),
250  .flush = libgsm_flush,
251  .capabilities = CODEC_CAP_DR1,
252  .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
253 };
254 #endif
const char * s
Definition: avisynth_c.h:668
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define GSM_FRAME_SIZE
Definition: gsm.h:29
This structure describes decoded (raw) audio or video data.
Definition: frame.h:76
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
signed 16 bits
Definition: samplefmt.h:52
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
initialize output if(nPeaks >3)%at least 3 peaks in spectrum for trying to find f0 nf0peaks
enum AVSampleFormat sample_fmt
audio sample format
uint8_t
#define av_cold
Definition: attributes.h:78
#define GSM_MS_BLOCK_SIZE
Definition: gsm.h:26
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
#define GSM_BLOCK_SIZE
Definition: gsm.h:25
uint8_t * data
frame
Definition: stft.m:14
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Spectrum Plot time data
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:246
const char * name
Name of the codec implementation.
external API header
uint64_t channel_layout
Audio channel layout.
static av_cold int libgsm_encode_close(AVCodecContext *avctx)
Definition: libgsm.c:43
int bit_rate
the average bitrate
audio channel layout utility functions
ret
Definition: avfilter.c:821
static av_cold int libgsm_decode_close(AVCodecContext *avctx)
Definition: libgsm.c:176
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
static void flush(AVCodecContext *avctx)
struct gsm_state * state
Definition: libgsm.c:146
int frame_size
Number of samples per channel in an audio frame.
NULL
Definition: eval.c:55
#define FF_COMPLIANCE_UNOFFICIAL
Allow unofficial extensions.
enum AVCodecID codec_id
int sample_rate
samples per second
main external API structure.
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:375
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:148
void * buf
Definition: avisynth_c.h:594
synthesis window for stochastic i
static uint32_t state
Definition: trasher.c:27
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:87
static void libgsm_flush(AVCodecContext *avctx)
Definition: libgsm.c:217
common internal api header.
common internal and external API header
static av_cold int libgsm_encode_init(AVCodecContext *avctx)
Definition: libgsm.c:49
static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: libgsm.c:94
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
static av_cold int libgsm_decode_init(AVCodecContext *avctx)
Definition: libgsm.c:149
static int libgsm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: libgsm.c:184
as in Berlin toast format
int channels
number of audio channels
struct LibGSMDecodeContext LibGSMDecodeContext
Filter the word “frame” indicates either a video frame or a group of audio samples
static int decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *avpkt)
Definition: crystalhd.c:868
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:127
int strict_std_compliance
strictly follow the standard (MPEG4, ...).
for(j=16;j >0;--j)