FFmpeg
audio_convert.h
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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 AVRESAMPLE_AUDIO_CONVERT_H
22 #define AVRESAMPLE_AUDIO_CONVERT_H
23 
24 #include "libavutil/samplefmt.h"
25 #include "avresample.h"
26 #include "internal.h"
27 #include "audio_data.h"
28 
29 /**
30  * Set conversion function if the parameters match.
31  *
32  * This compares the parameters of the conversion function to the parameters
33  * in the AudioConvert context. If the parameters do not match, no changes are
34  * made to the active functions. If the parameters do match and the alignment
35  * is not constrained, the function is set as the generic conversion function.
36  * If the parameters match and the alignment is constrained, the function is
37  * set as the optimized conversion function.
38  *
39  * @param ac AudioConvert context
40  * @param out_fmt output sample format
41  * @param in_fmt input sample format
42  * @param channels number of channels, or 0 for any number of channels
43  * @param ptr_align buffer pointer alignment, in bytes
44  * @param samples_align buffer size alignment, in samples
45  * @param descr function type description (e.g. "C" or "SSE")
46  * @param conv conversion function pointer
47  */
49  enum AVSampleFormat in_fmt, int channels,
50  int ptr_align, int samples_align,
51  const char *descr, void *conv);
52 
53 /**
54  * Allocate and initialize AudioConvert context for sample format conversion.
55  *
56  * @param avr AVAudioResampleContext
57  * @param out_fmt output sample format
58  * @param in_fmt input sample format
59  * @param channels number of channels
60  * @param sample_rate sample rate (used for dithering)
61  * @param apply_map apply channel map during conversion
62  * @return newly-allocated AudioConvert context
63  */
65  enum AVSampleFormat out_fmt,
66  enum AVSampleFormat in_fmt,
67  int channels, int sample_rate,
68  int apply_map);
69 
70 /**
71  * Free AudioConvert.
72  *
73  * The AudioConvert must have been previously allocated with ff_audio_convert_alloc().
74  *
75  * @param ac AudioConvert struct
76  */
78 
79 /**
80  * Convert audio data from one sample format to another.
81  *
82  * For each call, the alignment of the input and output AudioData buffers are
83  * examined to determine whether to use the generic or optimized conversion
84  * function (when available).
85  *
86  * The number of samples to convert is determined by in->nb_samples. The output
87  * buffer must be large enough to handle this many samples. out->nb_samples is
88  * set by this function before a successful return.
89  *
90  * @param ac AudioConvert context
91  * @param out output audio data
92  * @param in input audio data
93  * @return 0 on success, negative AVERROR code on failure
94  */
96 
97 /* arch-specific initialization functions */
98 
102 
103 #endif /* AVRESAMPLE_AUDIO_CONVERT_H */
ff_audio_convert_free
void ff_audio_convert_free(AudioConvert **ac)
Free AudioConvert.
out
FILE * out
Definition: movenc.c:54
ff_audio_convert_set_func
void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int ptr_align, int samples_align, const char *descr, void *conv)
Set conversion function if the parameters match.
Definition: audio_convert.c:70
avresample.h
channels
channels
Definition: aptx.c:30
sample_rate
sample_rate
Definition: ffmpeg_filter.c:191
AudioData
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
AVAudioResampleContext
Definition: internal.h:53
samplefmt.h
ff_audio_convert_init_aarch64
void ff_audio_convert_init_aarch64(AudioConvert *ac)
Definition: audio_convert_init.c:34
ff_audio_convert_init_arm
void ff_audio_convert_init_arm(AudioConvert *ac)
Definition: audio_convert_init.c:34
conv
static int conv(int samples, float **pcm, char *buf, int channels)
Definition: libvorbisdec.c:131
internal.h
audio_data.h
ff_audio_convert_init_x86
void ff_audio_convert_init_x86(AudioConvert *ac)
Definition: audio_convert_init.c:146
in
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
Definition: audio_convert.c:326
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AudioConvert
Definition: audio_convert.c:48
ff_audio_convert_alloc
AudioConvert * ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map)
Allocate and initialize AudioConvert context for sample format conversion.
ff_audio_convert
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
Convert audio data from one sample format to another.