audiointerleave.h
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1 /*
2  * audio interleaving prototypes and declarations
3  *
4  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
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 AVFORMAT_AUDIOINTERLEAVE_H
24 #define AVFORMAT_AUDIOINTERLEAVE_H
25 
26 #include "libavutil/fifo.h"
27 #include "avformat.h"
28 
29 typedef struct AudioInterleaveContext {
31  unsigned fifo_size; ///< size of currently allocated FIFO
32  uint64_t dts; ///< current dts
33  int sample_size; ///< size of one sample all channels included
34  const int *samples_per_frame; ///< must be 0-terminated
35  const int *samples; ///< current samples per frame, pointer to samples_per_frame
36  AVRational time_base; ///< time base of output audio packets
38 
41 
42 /**
43  * Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame
44  * and interleave them correctly.
45  * The first element of AVStream->priv_data must be AudioInterleaveContext
46  * when using this function.
47  *
48  * @param get_packet function will output a packet when streams are correctly interleaved.
49  * @param compare_ts function will compare AVPackets and decide interleaving order.
50  */
52  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
53  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *));
54 
55 #endif /* AVFORMAT_AUDIOINTERLEAVE_H */
struct AudioInterleaveContext AudioInterleaveContext
const char * s
Definition: avisynth_c.h:668
const int * samples
current samples per frame, pointer to samples_per_frame
const int * samples_per_frame
must be 0-terminated
Format I/O context.
Definition: avformat.h:944
static AVPacket pkt
Definition: demuxing.c:56
unsigned fifo_size
size of currently allocated FIFO
int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base)
uint64_t dts
current dts
void ff_audio_interleave_close(AVFormatContext *s)
static void flush(AVCodecContext *avctx)
AVRational time_base
time base of output audio packets
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, int(*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), int(*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame and interleave them correctly...
a very simple circular buffer FIFO implementation
rational number numerator/denominator
Definition: rational.h:43
int sample_size
size of one sample all channels included
Main libavformat public API header.
static int get_packet(URLContext *s, int for_header)
Interact with the server by receiving and sending RTMP packets until there is some significant data (...
Definition: rtmpproto.c:2122
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=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;}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_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_dlog(ac->avr,"%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-> out
This structure stores compressed data.