FFmpeg  4.0
audioconvert.h
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1 /*
2  * audio conversion
3  * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
4  * Copyright (c) 2008 Peter Ross
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #ifndef SWRESAMPLE_AUDIOCONVERT_H
24 #define SWRESAMPLE_AUDIOCONVERT_H
25 
26 /**
27  * @file
28  * Audio format conversion routines
29  */
30 
31 
32 #include "swresample_internal.h"
33 #include "libavutil/cpu.h"
34 
35 
36 typedef void (conv_func_type)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end);
37 typedef void (simd_func_type)(uint8_t **dst, const uint8_t **src, int len);
38 
39 typedef struct AudioConvert {
40  int channels;
45  const int *ch_map;
46  uint8_t silence[8]; ///< silence input sample
48 
49 /**
50  * Create an audio sample format converter context
51  * @param out_fmt Output sample format
52  * @param in_fmt Input sample format
53  * @param channels Number of channels
54  * @param flags See AV_CPU_FLAG_xx
55  * @param ch_map list of the channels id to pick from the source stream, NULL
56  * if all channels must be selected
57  * @return NULL on error
58  */
61  int channels, const int *ch_map,
62  int flags);
63 
64 /**
65  * Free audio sample format converter context.
66  * and set the pointer to NULL
67  */
69 
70 /**
71  * Convert between audio sample formats
72  * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
73  * @param[in] in array of input buffers for each channel
74  * @param len length of audio frame size (measured in samples)
75  */
77 
78 #endif /* SWRESAMPLE_AUDIOCONVERT_H */
simd_func_type * simd_f
Definition: audioconvert.h:44
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
channels
Definition: aptx.c:30
#define src
Definition: vp8dsp.c:254
uint8_t
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int in_simd_align_mask
Definition: audioconvert.h:41
static int flags
Definition: log.c:55
int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len)
Convert between audio sample formats.
void() simd_func_type(uint8_t **dst, const uint8_t **src, int len)
Definition: audioconvert.h:37
AVFormatContext * ctx
Definition: movenc.c:48
void() conv_func_type(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)
Definition: audioconvert.h:36
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
typedef void(RENAME(mix_any_func_type))
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
int out_simd_align_mask
Definition: audioconvert.h:42
uint8_t silence[8]
silence input sample
Definition: audioconvert.h:46
conv_func_type * conv_f
Definition: audioconvert.h:43
void swri_audio_convert_free(AudioConvert **ctx)
Free audio sample format converter context.
int len
enum AVSampleFormat out_fmt
Definition: audio_convert.c:52
enum AVSampleFormat in_fmt
Definition: audio_convert.c:51
FILE * out
Definition: movenc.c:54
const int * ch_map
Definition: audioconvert.h:45
AudioConvert * swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags)
Create an audio sample format converter context.