vorbis.h
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1 /*
2  * copyright (c) 2006 Oded Shimon <ods15@ods15.dyndns.org>
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 #ifndef AVCODEC_VORBIS_H
22 #define AVCODEC_VORBIS_H
23 
24 #include "avcodec.h"
25 
26 extern const float ff_vorbis_floor1_inverse_db_table[256];
27 extern const float * const ff_vorbis_vwin[8];
28 extern const uint8_t ff_vorbis_channel_layout_offsets[8][8];
29 extern const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8];
30 extern const uint64_t ff_vorbis_channel_layouts[9];
31 
32 typedef struct vorbis_floor1_entry {
33  uint16_t x;
34  uint16_t sort;
35  uint16_t low;
36  uint16_t high;
37 } vorbis_floor1_entry;
38 
39 int ff_vorbis_ready_floor1_list(AVCodecContext *avctx,
40  vorbis_floor1_entry *list, int values);
41 unsigned int ff_vorbis_nth_root(unsigned int x, unsigned int n); // x^(1/n)
42 int ff_vorbis_len2vlc(uint8_t *bits, uint32_t *codes, unsigned num);
43 void ff_vorbis_floor1_render_list(vorbis_floor1_entry * list, int values,
44  uint16_t *y_list, int *flag,
45  int multiplier, float * out, int samples);
46 void ff_vorbis_inverse_coupling(float *mag, float *ang, intptr_t blocksize);
47 
48 #define ilog(i) av_log2(2*(i))
49 
50 #endif /* AVCODEC_VORBIS_H */
ff_vorbis_ready_floor1_list
int ff_vorbis_ready_floor1_list(AVCodecContext *avctx, vorbis_floor1_entry *list, int values)
Definition: vorbis.c:124
ff_vorbis_floor1_render_list
void ff_vorbis_floor1_render_list(vorbis_floor1_entry *list, int values, uint16_t *y_list, int *flag, int multiplier, float *out, int samples)
Definition: vorbis.c:214
ff_vorbis_floor1_inverse_db_table
const float ff_vorbis_floor1_inverse_db_table[256]
Definition: vorbis_data.c:2123
vorbis_floor1_entry::sort
uint16_t sort
Definition: vorbis.h:34
list
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer 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 list
Definition: filter_design.txt:23
bits
uint8_t bits
Definition: crc.c:216
uint8_t
uint8_t
Definition: audio_convert.c:194
vorbis_floor1_entry::x
uint16_t x
Definition: vorbis.h:33
avcodec.h
external API header
vorbis_floor1_entry
struct vorbis_floor1_entry vorbis_floor1_entry
ff_vorbis_encoding_channel_layout_offsets
const uint8_t ff_vorbis_encoding_channel_layout_offsets[8][8]
Definition: vorbis_data.c:36
AVCodecContext
main external API structure.
Definition: libavcodec/avcodec.h:1134
ff_vorbis_channel_layouts
const uint64_t ff_vorbis_channel_layouts[9]
Definition: vorbis_data.c:47
vorbis_floor1_entry::low
uint16_t low
Definition: vorbis.h:35
mag
mag
Definition: lab5.m:14
ff_vorbis_len2vlc
int ff_vorbis_len2vlc(uint8_t *bits, uint32_t *codes, unsigned num)
Definition: vorbis.c:57
vorbis_floor1_entry::high
uint16_t high
Definition: vorbis.h:36
ff_vorbis_vwin
const float *const ff_vorbis_vwin[8]
Definition: vorbis_data.c:2190
n
n
Definition: genspecsines_sin.m:10
ff_vorbis_inverse_coupling
void ff_vorbis_inverse_coupling(float *mag, float *ang, intptr_t blocksize)
Definition: vorbisdec.c:1491
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:2
out
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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_dlog(ac->avr,"%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-> out
Definition: audio_convert.c:194
ff_vorbis_channel_layout_offsets
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
Definition: vorbis_data.c:25
vorbis_floor1_entry
Definition: vorbis.h:32
ff_vorbis_nth_root
unsigned int ff_vorbis_nth_root(unsigned int x, unsigned int n)
Definition: vorbis.c:39
values
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 This method is called when a frame is wanted on an output For an 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 values
Definition: filter_design.txt:216
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