1 /*
2 * Copyright (c) 2014 Luca Barbato <lu_zero@gentoo.org>
3 * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
25
27 {
29
30 if (in) {
37 }
38
39 if (out) {
46 }
47
48 return 0;
52 }
53
56 {
57 int ret = 0;
58
59 if (in) {
64 }
65 }
66
67 if (out) {
72 }
73 }
74
75 return ret;
76 }
77
80 {
81 int ret;
84 int out_nb_samples = 0, in_nb_samples = 0;
85
86 if (out) {
89 }
90
91 if (in) {
94 }
95
96 ret =
swr_convert(s, out_data, out_nb_samples, in_data, in_nb_samples);
97
98 if (ret < 0) {
99 if (out)
101 return ret;
102 }
103
104 if (out)
106
107 return 0;
108 }
109
111 {
113 int samples = out->
linesize[0] / bytes_per_sample;
114
116 return samples;
117 } else {
119 return samples / channels;
120 }
121 }
122
125 {
126 int ret, setup = 0;
127
130 return ret;
132 return ret;
133 setup = 1;
134 } else {
135 // return as is or reconfigure for input changes?
137 return ret;
138 }
139
140 if (out) {
144 + 3;
146 if (setup)
148 return ret;
149 }
150 } else {
153 }
154 }
155
157 }
158
av_cold void swr_close(SwrContext *s)
Closes the context so that swr_is_initialized() returns 0.
int out_sample_rate
output sample rate
This structure describes decoded (raw) audio or video data.
static int config_changed(SwrContext *s, const AVFrame *out, const AVFrame *in)
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
enum AVSampleFormat out_sample_fmt
output sample format
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
The libswresample context.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
reference-counted frame API
uint64_t channel_layout
Channel layout of the audio data.
static int convert_frame(SwrContext *s, AVFrame *out, const AVFrame *in)
int64_t out_ch_layout
output channel layout
#define AVERROR_INPUT_CHANGED
Input changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_OUTPUT_CHANGED) ...
int in_sample_rate
input sample rate
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
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
int sample_rate
Sample rate of the audio data.
enum AVSampleFormat in_sample_fmt
input sample format
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
#define AVERROR_OUTPUT_CHANGED
Output changed between calls. Reconfiguration is required. (can be OR-ed with AVERROR_INPUT_CHANGED) ...
int64_t in_ch_layout
input channel layout
int swr_convert_frame(SwrContext *s, AVFrame *out, const AVFrame *in)
Convert the samples in the input AVFrame and write them to the output AVFrame.
static int available_samples(AVFrame *out)
int swr_config_frame(SwrContext *s, const AVFrame *out, const AVFrame *in)
Configure or reconfigure the SwrContext using the information provided by the AVFrames.
uint8_t ** extended_data
pointers to the data planes/channels.
int swr_is_initialized(struct SwrContext *s)
Check whether an swr context has been initialized or not.
int nb_samples
number of audio samples (per channel) described by this frame
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.