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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 AVCODEC_AUDIOCONVERT_H
24 #define AVCODEC_AUDIOCONVERT_H
25 
26 #include "version.h"
27 
28 /**
29  * @file
30  * Audio format conversion routines
31  * This interface is deprecated and will be dropped in a future
32  * version. You should use the libswresample library instead.
33  */
34 
35 #if FF_API_AUDIO_CONVERT
36 
37 #include "libavutil/cpu.h"
38 #include "avcodec.h"
40 
41 struct AVAudioConvert;
42 typedef struct AVAudioConvert AVAudioConvert;
43 
44 /**
45  * Create an audio sample format converter context
46  * @param out_fmt Output sample format
47  * @param out_channels Number of output channels
48  * @param in_fmt Input sample format
49  * @param in_channels Number of input channels
50  * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
51  * @param flags See AV_CPU_FLAG_xx
52  * @return NULL on error
53  * @deprecated See libswresample
54  */
55 
58  enum AVSampleFormat in_fmt, int in_channels,
59  const float *matrix, int flags);
60 
61 /**
62  * Free audio sample format converter context
63  * @deprecated See libswresample
64  */
65 
68 
69 /**
70  * Convert between audio sample formats
71  * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
72  * @param[in] out_stride distance between consecutive output samples (measured in bytes)
73  * @param[in] in array of input buffers for each channel
74  * @param[in] in_stride distance between consecutive input samples (measured in bytes)
75  * @param len length of audio frame size (measured in samples)
76  * @deprecated See libswresample
77  */
78 
81  void * const out[6], const int out_stride[6],
82  const void * const in[6], const int in_stride[6], int len);
83 
84 #endif /* FF_API_AUDIO_CONVERT */
85 
86 #endif /* AVCODEC_AUDIOCONVERT_H */
static int flags
Definition: log.c:57
Libavcodec version macros.
attribute_deprecated void av_audio_convert_free(AVAudioConvert *ctx)
Free audio sample format converter context.
Definition: audioconvert.c:58
audio channel layout utility functions
AVFormatContext * ctx
Definition: movenc.c:48
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
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
#define attribute_deprecated
Definition: attributes.h:94
attribute_deprecated int av_audio_convert(AVAudioConvert *ctx, void *const out[6], const int out_stride[6], const void *const in[6], const int in_stride[6], int len)
Convert between audio sample formats.
Definition: audioconvert.c:63
int len
attribute_deprecated AVAudioConvert * av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels, enum AVSampleFormat in_fmt, int in_channels, const float *matrix, int flags)
Create an audio sample format converter context.
Definition: audioconvert.c:42
FILE * out
Definition: movenc.c:54