FFmpeg
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_unref(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:5765
int size
Definition: avcodec.h:1480
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:5815
uint8_t * data
Definition: avcodec.h:1479
void av_packet_move_ref(AVPacket *dst, AVPacket *src)
Move every field in src to dst and reset src.
Definition: avpacket.c:655
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:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:566
static int ff_mpa_check_header(uint32_t header)
#define fail()
Definition: checkasm.h:121
int extradata_size
Size of the extradata content in bytes.
Definition: avcodec.h:3977
int av_packet_copy_props(AVPacket *dst, const AVPacket *src)
Copy only "properties" fields from src to dst.
Definition: avpacket.c:565
AVFormatContext * ctx
Definition: movenc.c:48
sample_rate
int frame_size
Definition: mxfenc.c:2214
Libavcodec external API header.
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:599
void * buf
Definition: avisynth_c.h:766
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;}return ac;}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;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
#define AV_WB32(p, v)
Definition: intreadwrite.h:419
int sample_rate
Audio only.
Definition: avcodec.h:4069
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:216
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:792
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:3973
int channels
Audio only.
Definition: avcodec.h:4065
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
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
This structure stores compressed data.
Definition: avcodec.h:1456
AVCodecParameters * par_in
Parameters of the input stream.
Definition: avcodec.h:5793
const AVBitStreamFilter ff_mp3_header_decompress_bsf