FFmpeg
libvorbisdec.c
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1 /*
2  * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include <vorbis/vorbisenc.h>
22 
23 #include "avcodec.h"
24 #include "bytestream.h"
25 #include "internal.h"
26 
27 typedef struct OggVorbisDecContext {
28  vorbis_info vi; /**< vorbis_info used during init */
29  vorbis_dsp_state vd; /**< DSP state used for analysis */
30  vorbis_block vb; /**< vorbis_block used for analysis */
31  vorbis_comment vc; /**< VorbisComment info */
32  ogg_packet op; /**< ogg packet */
34 
35 static int oggvorbis_decode_close(AVCodecContext *avccontext);
36 
37 static int oggvorbis_decode_init(AVCodecContext *avccontext) {
38  OggVorbisDecContext *context = avccontext->priv_data ;
39  uint8_t *p= avccontext->extradata;
40  int i, hsizes[3], ret;
41  unsigned char *headers[3], *extradata = avccontext->extradata;
42 
43  if(! avccontext->extradata_size || ! p) {
44  av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n");
45  return AVERROR(EINVAL);
46  }
47 
48  vorbis_info_init(&context->vi) ;
49  vorbis_comment_init(&context->vc) ;
50 
51  if(p[0] == 0 && p[1] == 30) {
52  int sizesum = 0;
53  for(i = 0; i < 3; i++){
54  hsizes[i] = bytestream_get_be16((const uint8_t **)&p);
55  sizesum += 2 + hsizes[i];
56  if (sizesum > avccontext->extradata_size) {
57  av_log(avccontext, AV_LOG_ERROR, "vorbis extradata too small\n");
58  ret = AVERROR_INVALIDDATA;
59  goto error;
60  }
61 
62  headers[i] = p;
63  p += hsizes[i];
64  }
65  } else if(*p == 2) {
66  unsigned int offset = 1;
67  p++;
68  for(i=0; i<2; i++) {
69  hsizes[i] = 0;
70  while((*p == 0xFF) && (offset < avccontext->extradata_size)) {
71  hsizes[i] += 0xFF;
72  offset++;
73  p++;
74  }
75  if(offset >= avccontext->extradata_size - 1) {
76  av_log(avccontext, AV_LOG_ERROR,
77  "vorbis header sizes damaged\n");
78  ret = AVERROR_INVALIDDATA;
79  goto error;
80  }
81  hsizes[i] += *p;
82  offset++;
83  p++;
84  }
85  hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset;
86 #if 0
87  av_log(avccontext, AV_LOG_DEBUG,
88  "vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
89  hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size);
90 #endif
91  headers[0] = extradata + offset;
92  headers[1] = extradata + offset + hsizes[0];
93  headers[2] = extradata + offset + hsizes[0] + hsizes[1];
94  } else {
95  av_log(avccontext, AV_LOG_ERROR,
96  "vorbis initial header len is wrong: %d\n", *p);
97  ret = AVERROR_INVALIDDATA;
98  goto error;
99  }
100 
101  for(i=0; i<3; i++){
102  context->op.b_o_s= i==0;
103  context->op.bytes = hsizes[i];
104  context->op.packet = headers[i];
105  if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){
106  av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1);
107  ret = AVERROR_INVALIDDATA;
108  goto error;
109  }
110  }
111 
112  avccontext->channels = context->vi.channels;
113  avccontext->sample_rate = context->vi.rate;
114  avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
115  avccontext->time_base= (AVRational){1, avccontext->sample_rate};
116 
117  vorbis_synthesis_init(&context->vd, &context->vi);
118  vorbis_block_init(&context->vd, &context->vb);
119 
120  return 0 ;
121 
122  error:
123  oggvorbis_decode_close(avccontext);
124  return ret;
125 }
126 
127 
128 static inline int conv(int samples, float **pcm, char *buf, int channels) {
129  int i, j;
130  ogg_int16_t *ptr, *data = (ogg_int16_t*)buf ;
131  float *mono ;
132 
133  for(i = 0 ; i < channels ; i++){
134  ptr = &data[i];
135  mono = pcm[i] ;
136 
137  for(j = 0 ; j < samples ; j++) {
138  *ptr = av_clip_int16(mono[j] * 32767.f);
139  ptr += channels;
140  }
141  }
142 
143  return 0 ;
144 }
145 
146 static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data,
147  int *got_frame_ptr, AVPacket *avpkt)
148 {
149  OggVorbisDecContext *context = avccontext->priv_data ;
150  AVFrame *frame = data;
151  float **pcm ;
152  ogg_packet *op= &context->op;
153  int samples, total_samples, total_bytes;
154  int ret;
155  int16_t *output;
156 
157  if(!avpkt->size){
158  //FIXME flush
159  return 0;
160  }
161 
162  frame->nb_samples = 8192*4;
163  if ((ret = ff_get_buffer(avccontext, frame, 0)) < 0)
164  return ret;
165  output = (int16_t *)frame->data[0];
166 
167 
168  op->packet = avpkt->data;
169  op->bytes = avpkt->size;
170 
171 // av_log(avccontext, AV_LOG_DEBUG, "%d %d %d %"PRId64" %"PRId64" %d %d\n", op->bytes, op->b_o_s, op->e_o_s, op->granulepos, op->packetno, buf_size, context->vi.rate);
172 
173 /* for(i=0; i<op->bytes; i++)
174  av_log(avccontext, AV_LOG_DEBUG, "%02X ", op->packet[i]);
175  av_log(avccontext, AV_LOG_DEBUG, "\n");*/
176 
177  if(vorbis_synthesis(&context->vb, op) == 0)
178  vorbis_synthesis_blockin(&context->vd, &context->vb) ;
179 
180  total_samples = 0 ;
181  total_bytes = 0 ;
182 
183  while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) {
184  conv(samples, pcm, (char*)output + total_bytes, context->vi.channels) ;
185  total_bytes += samples * 2 * context->vi.channels ;
186  total_samples += samples ;
187  vorbis_synthesis_read(&context->vd, samples) ;
188  }
189 
190  frame->nb_samples = total_samples;
191  *got_frame_ptr = total_samples > 0;
192  return avpkt->size;
193 }
194 
195 
196 static int oggvorbis_decode_close(AVCodecContext *avccontext) {
197  OggVorbisDecContext *context = avccontext->priv_data ;
198 
199  vorbis_block_clear(&context->vb);
200  vorbis_dsp_clear(&context->vd);
201  vorbis_info_clear(&context->vi) ;
202  vorbis_comment_clear(&context->vc) ;
203 
204  return 0 ;
205 }
206 
207 
209  .name = "libvorbis",
210  .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
211  .type = AVMEDIA_TYPE_AUDIO,
212  .id = AV_CODEC_ID_VORBIS,
213  .priv_data_size = sizeof(OggVorbisDecContext),
216  .close = oggvorbis_decode_close,
217  .capabilities = AV_CODEC_CAP_DELAY,
218 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
ogg_packet op
ogg packet
Definition: libvorbisdec.c:32
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
static int conv(int samples, float **pcm, char *buf, int channels)
Definition: libvorbisdec.c:128
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
vorbis_dsp_state vd
DSP state used for analysis.
Definition: libvorbisdec.c:29
channels
Definition: aptx.c:30
int size
Definition: avcodec.h:1478
AVCodec.
Definition: avcodec.h:3481
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avcodec.h:1688
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:1006
vorbis_block vb
vorbis_block used for analysis
Definition: libvorbisdec.c:30
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
uint8_t
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
#define f(width, name)
Definition: cbs_vp9.c:255
static int oggvorbis_decode_init(AVCodecContext *avccontext)
Definition: libvorbisdec.c:37
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
uint8_t * data
Definition: avcodec.h:1477
#define av_log(a,...)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
vorbis_comment vc
VorbisComment info.
Definition: libvorbisdec.c:31
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
vorbis_info vi
vorbis_info used during init
Definition: libvorbisdec.c:28
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
static void error(const char *err)
Libavcodec external API header.
static int oggvorbis_decode_frame(AVCodecContext *avccontext, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: libvorbisdec.c:146
int sample_rate
samples per second
Definition: avcodec.h:2225
static int ogg_packet(AVFormatContext *s, int *sid, int *dstart, int *dsize, int64_t *fpos)
find the next Ogg packet
Definition: oggdec.c:477
main external API structure.
Definition: avcodec.h:1565
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1964
static int oggvorbis_decode_close(AVCodecContext *avccontext)
Definition: libvorbisdec.c:196
void * buf
Definition: avisynth_c.h:766
int extradata_size
Definition: avcodec.h:1667
Rational number (pair of numerator and denominator).
Definition: rational.h:58
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
AVCodec ff_libvorbis_decoder
Definition: libvorbisdec.c:208
void * priv_data
Definition: avcodec.h:1592
int channels
number of audio channels
Definition: avcodec.h:2226
Filter the word “frame” indicates either a video frame or a group of audio samples
FFmpeg currently uses a custom build this text attempts to document some of its obscure features and options Makefile the full command issued by make and its output will be shown on the screen DBG Preprocess x86 external assembler files to a dbg asm file in the object which then gets compiled Helps in developing those assembler files DESTDIR Destination directory for the install useful to prepare packages or install FFmpeg in cross environments GEN Set to ‘1’ to generate the missing or mismatched references Makefile builds all the libraries and the executables fate Run the fate test note that you must have installed it fate list List all fate regression test targets install Install headers
Definition: build_system.txt:1
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option keep it simple and lowercase description are in without and describe what they for example set the foo of the bar offset is the offset of the field in your local context
This structure stores compressed data.
Definition: avcodec.h:1454
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361