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mp3_header_decompress_bsf.c
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1 /*
2  * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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 "libavutil/common.h"
22 #include "libavutil/intreadwrite.h"
23 #include "avcodec.h"
24 #include "bsf.h"
25 #include "mpegaudiodecheader.h"
26 #include "mpegaudiodata.h"
27 
28 
30 {
31  AVPacket *in;
32  uint32_t header;
33  int sample_rate= ctx->par_in->sample_rate;
34  int sample_rate_index=0;
35  int lsf, mpeg25, bitrate_index, frame_size, ret;
36  uint8_t *buf;
37  int buf_size;
38 
39  ret = ff_bsf_get_packet(ctx, &in);
40  if (ret < 0)
41  return ret;
42 
43  buf = in->data;
44  buf_size = in->size;
45 
46  header = AV_RB32(buf);
47  if(ff_mpa_check_header(header) >= 0){
48  av_packet_move_ref(out, in);
49  av_packet_free(&in);
50 
51  return 0;
52  }
53 
54  if(ctx->par_in->extradata_size != 15 || strcmp(ctx->par_in->extradata, "FFCMP3 0.0")){
55  av_log(ctx, AV_LOG_ERROR, "Extradata invalid %d\n", ctx->par_in->extradata_size);
56  ret = AVERROR(EINVAL);
57  goto fail;
58  }
59 
60  header= AV_RB32(ctx->par_in->extradata+11) & MP3_MASK;
61 
62  lsf = sample_rate < (24000+32000)/2;
63  mpeg25 = sample_rate < (12000+16000)/2;
64  sample_rate_index= (header>>10)&3;
65  sample_rate= avpriv_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
66 
67  for(bitrate_index=2; bitrate_index<30; bitrate_index++){
68  frame_size = avpriv_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
69  frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
70  if(frame_size == buf_size + 4)
71  break;
72  if(frame_size == buf_size + 6)
73  break;
74  }
75  if(bitrate_index == 30){
76  av_log(ctx, AV_LOG_ERROR, "Could not find bitrate_index.\n");
77  ret = AVERROR(EINVAL);
78  goto fail;
79  }
80 
81  header |= (bitrate_index&1)<<9;
82  header |= (bitrate_index>>1)<<12;
83  header |= (frame_size == buf_size + 4)<<16; //FIXME actually set a correct crc instead of 0
84 
85  ret = av_new_packet(out, frame_size);
86  if (ret < 0)
87  goto fail;
88  ret = av_packet_copy_props(out, in);
89  if (ret < 0) {
90  av_packet_free(&out);
91  goto fail;
92  }
93  memcpy(out->data + frame_size - buf_size, buf, buf_size + AV_INPUT_BUFFER_PADDING_SIZE);
94 
95  if(ctx->par_in->channels==2){
96  uint8_t *p= out->data + frame_size - buf_size;
97  if(lsf){
98  FFSWAP(int, p[1], p[2]);
99  header |= (p[1] & 0xC0)>>2;
100  p[1] &= 0x3F;
101  }else{
102  header |= p[1] & 0x30;
103  p[1] &= 0xCF;
104  }
105  }
106 
107  AV_WB32(out->data, header);
108 
109  ret = 0;
110 
111 fail:
112  av_packet_free(&in);
113  return ret;
114 }
115 
116 static const enum AVCodecID codec_ids[] = {
118 };
119 
121  .name = "mp3decomp",
122  .filter = mp3_header_decompress,
123  .codec_ids = codec_ids,
124 };
#define MP3_MASK
The bitstream filter state.
Definition: avcodec.h:5732
int size
Definition: avcodec.h:1602
mpeg audio layer common tables.
static enum AVCodecID codec_ids[]
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:62
uint8_t
const uint16_t avpriv_mpa_freq_tab[3]
Definition: mpegaudiodata.c:40
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
const char * name
Definition: avcodec.h:5779
uint8_t * data
Definition: avcodec.h:1601
void av_packet_move_ref(AVPacket *dst, AVPacket *src)
Move every field in src to dst and reset src.
Definition: avpacket.c:623
static const uint8_t header[24]
Definition: sdr2.c:67
#define av_log(a,...)
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:86
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:191
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:517
static int ff_mpa_check_header(uint32_t header)
#define fail()
Definition: checkasm.h:84
int extradata_size
Size of the extradata content in bytes.
Definition: avcodec.h:3999
int av_packet_copy_props(AVPacket *dst, const AVPacket *src)
Copy only "properties" fields from src to dst.
Definition: avpacket.c:536
AVFormatContext * ctx
Definition: movenc.c:48
sample_rate
int frame_size
Definition: mxfenc.c:1820
Libavcodec external API header.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
void * buf
Definition: avisynth_c.h:690
#define AV_WB32(p, v)
Definition: intreadwrite.h:419
int sample_rate
Audio only.
Definition: avcodec.h:4091
MPEG Audio header decoder.
common internal and external API header
int ff_bsf_get_packet(AVBSFContext *ctx, AVPacket **pkt)
Called by the bitstream filters to get the next packet for filtering.
Definition: bsf.c:204
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:734
static int mp3_header_decompress(AVBSFContext *ctx, AVPacket *out)
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: avcodec.h:3995
int channels
Audio only.
Definition: avcodec.h:4087
FILE * out
Definition: movenc.c:54
const uint16_t avpriv_mpa_bitrate_tab[2][3][15]
Definition: mpegaudiodata.c:30
#define FFSWAP(type, a, b)
Definition: common.h:99
This structure stores compressed data.
Definition: avcodec.h:1578
AVCodecParameters * par_in
Parameters of the input stream.
Definition: avcodec.h:5758
const AVBitStreamFilter ff_mp3_header_decompress_bsf