Libav
audio_data.h
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1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3  *
4  * This file is part of Libav.
5  *
6  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef AVRESAMPLE_AUDIO_DATA_H
22 #define AVRESAMPLE_AUDIO_DATA_H
23 
24 #include <stdint.h>
25 
26 #include "libavutil/audio_fifo.h"
27 #include "libavutil/log.h"
28 #include "libavutil/samplefmt.h"
29 #include "avresample.h"
30 #include "internal.h"
31 
35 struct AudioData {
36  const AVClass *class;
39  unsigned int buffer_size;
41  int nb_samples;
43  int channels;
45  int is_planar;
46  int planes;
48  int stride;
49  int read_only;
51  int ptr_align;
53  const char *name;
54 };
55 
56 int ff_audio_data_set_channels(AudioData *a, int channels);
57 
77 int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
78  int nb_samples, enum AVSampleFormat sample_fmt,
79  int read_only, const char *name);
80 
92 AudioData *ff_audio_data_alloc(int channels, int nb_samples,
94  const char *name);
95 
105 int ff_audio_data_realloc(AudioData *a, int nb_samples);
106 
115 
125 
138 int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
139  int src_offset, int nb_samples);
140 
149 void ff_audio_data_drain(AudioData *a, int nb_samples);
150 
161 int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
162  int nb_samples);
163 
173 int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
174 
175 #endif /* AVRESAMPLE_AUDIO_DATA_H */
unsigned int buffer_size
allocated buffer size
Definition: audio_data.h:39
const char * name
name for debug logging
Definition: audio_data.h:53
void ff_audio_data_free(AudioData **a)
Free AudioData.
Definition: audio_data.c:208
Audio buffer used for intermediate storage between conversion phases.
Definition: oss_audio.c:47
int allow_realloc
realloc is allowed
Definition: audio_data.h:50
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, int src_offset, int nb_samples)
Append data from one AudioData to the end of another.
Definition: audio_data.c:269
int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map)
Copy data from one AudioData to another.
Definition: audio_data.c:216
int nb_samples
current number of samples
Definition: audio_data.h:41
int ff_audio_data_set_channels(AudioData *a, int channels)
Definition: audio_data.c:51
int sample_size
bytes per sample
Definition: audio_data.h:47
int allocated_channels
allocated channel count
Definition: audio_data.h:44
uint8_t
int ff_audio_data_realloc(AudioData *a, int nb_samples)
Reallocate AudioData.
Definition: audio_data.c:153
const char * name
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=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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%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-> out
int read_only
data is read-only
Definition: audio_data.h:49
void ff_audio_data_drain(AudioData *a, int nb_samples)
Drain samples from the start of the AudioData.
Definition: audio_data.c:325
int stride
sample byte offset within a plane
Definition: audio_data.h:48
int channels
channel count
Definition: oss_audio.c:51
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=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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%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
int is_planar
sample format is planar
Definition: audio_data.h:45
uint8_t * buffer
data buffer
Definition: audio_data.h:38
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
AudioData * ff_audio_data_alloc(int channels, int nb_samples, enum AVSampleFormat sample_fmt, const char *name)
Allocate AudioData.
Definition: audio_data.c:110
external API header
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
Read samples from an AVAudioFifo to AudioData.
Definition: audio_data.c:357
sample_fmt
Definition: avconv_filter.c:68
#define AVRESAMPLE_MAX_CHANNELS
Definition: avresample.h:103
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, int nb_samples)
Add samples in AudioData to an AVAudioFifo.
Definition: audio_data.c:342
Describe the class of an AVClass context structure.
Definition: log.h:33
int samples_align
allocated samples alignment
Definition: audio_data.h:52
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
Audio FIFO Buffer.
int allocated_samples
number of samples the buffer can hold
Definition: audio_data.h:40
enum AVSampleFormat sample_fmt
sample format
Definition: audio_data.h:42
int ptr_align
minimum data pointer alignment
Definition: audio_data.h:51
int planes
number of data planes
Definition: audio_data.h:46
int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, int nb_samples, enum AVSampleFormat sample_fmt, int read_only, const char *name)
Initialize AudioData using a given source.
Definition: audio_data.c:65
uint8_t * data[AVRESAMPLE_MAX_CHANNELS]
data plane pointers
Definition: audio_data.h:37