mirror of https://git.ffmpeg.org/ffmpeg.git
460 lines
16 KiB
C
460 lines
16 KiB
C
/*
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* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
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*
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* This file is part of libswresample
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*
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* libswresample is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* libswresample is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with libswresample; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "swresample_internal.h"
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#include "audioconvert.h"
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#include "libavutil/avassert.h"
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#include "libavutil/audioconvert.h"
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#define C30DB M_SQRT2
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#define C15DB 1.189207115
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#define C__0DB 1.0
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#define C_15DB 0.840896415
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#define C_30DB M_SQRT1_2
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#define C_45DB 0.594603558
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#define C_60DB 0.5
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//TODO split options array out?
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#define OFFSET(x) offsetof(SwrContext,x)
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static const AVOption options[]={
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{"ich", "input channel count", OFFSET( in.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
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{"och", "output channel count", OFFSET(out.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
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{"isr", "input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
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{"osr", "output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
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//{"ip" , "input planar" , OFFSET( in.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
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//{"op" , "output planar" , OFFSET(out.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
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{"isf", "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
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{"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
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{"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
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{"icl", "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
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{"ocl", "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
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{"clev", "center mix level" , OFFSET(clev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
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{"slev", "sourround mix level" , OFFSET(slev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
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{"flags", NULL , OFFSET(flags) , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"},
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{"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
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{0}
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};
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static const char* context_to_name(void* ptr) {
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return "SWR";
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}
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static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
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static int resample(SwrContext *s, AudioData *out_param, int out_count,
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const AudioData * in_param, int in_count);
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SwrContext *swr_alloc(void){
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SwrContext *s= av_mallocz(sizeof(SwrContext));
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if(s){
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s->av_class= &av_class;
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av_opt_set_defaults2(s, 0, 0);
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}
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return s;
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}
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SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
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int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
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int log_offset, void *log_ctx){
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if(!s) s= swr_alloc();
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if(!s) return NULL;
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s->log_level_offset= log_offset;
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s->log_ctx= log_ctx;
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av_set_int(s, "ocl", out_ch_layout);
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av_set_int(s, "osf", out_sample_fmt);
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av_set_int(s, "osr", out_sample_rate);
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av_set_int(s, "icl", in_ch_layout);
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av_set_int(s, "isf", in_sample_fmt);
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av_set_int(s, "isr", in_sample_rate);
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s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
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s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
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s->int_sample_fmt = AV_SAMPLE_FMT_S16;
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return s;
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}
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static void free_temp(AudioData *a){
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av_free(a->data);
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memset(a, 0, sizeof(*a));
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}
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void swr_free(SwrContext **ss){
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SwrContext *s= *ss;
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if(s){
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free_temp(&s->postin);
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free_temp(&s->midbuf);
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free_temp(&s->preout);
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free_temp(&s->in_buffer);
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swr_audio_convert_free(&s-> in_convert);
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swr_audio_convert_free(&s->out_convert);
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swr_audio_convert_free(&s->full_convert);
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swr_resample_free(&s->resample);
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}
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av_freep(ss);
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}
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int swr_init(SwrContext *s){
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s->in_buffer_index= 0;
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s->in_buffer_count= 0;
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s->resample_in_constraint= 0;
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free_temp(&s->postin);
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free_temp(&s->midbuf);
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free_temp(&s->preout);
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free_temp(&s->in_buffer);
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swr_audio_convert_free(&s-> in_convert);
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swr_audio_convert_free(&s->out_convert);
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swr_audio_convert_free(&s->full_convert);
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s-> in.planar= s-> in_sample_fmt >= 0x100;
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s->out.planar= s->out_sample_fmt >= 0x100;
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s-> in_sample_fmt &= 0xFF;
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s->out_sample_fmt &= 0xFF;
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if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
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av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
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return AVERROR(EINVAL);
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}
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if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
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av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
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return AVERROR(EINVAL);
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}
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
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&&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
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av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
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return AVERROR(EINVAL);
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}
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//FIXME should we allow/support using FLT on material that doesnt need it ?
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if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
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s->int_sample_fmt= AV_SAMPLE_FMT_S16;
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}else
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s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
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if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
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s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
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}else
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swr_resample_free(&s->resample);
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if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
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av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
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return -1;
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}
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if(s-> in.ch_count && s-> in_ch_layout && s->in.ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
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av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than there actually is, ignoring layout\n");
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s-> in_ch_layout= 0;
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}
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if(!s-> in_ch_layout)
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s-> in_ch_layout= av_get_default_channel_layout(s->in.ch_count);
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if(!s->out_ch_layout)
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s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
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s->rematrix= s->out_ch_layout !=s->in_ch_layout;
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#define RSC 1 //FIXME finetune
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if(!s-> in.ch_count)
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s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
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if(!s->out.ch_count)
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s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
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av_assert0(s-> in.ch_count);
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av_assert0(s->out.ch_count);
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s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
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s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
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s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
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s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
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if(!s->resample && !s->rematrix){
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s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
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s-> in_sample_fmt, s-> in.ch_count, 0);
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return 0;
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}
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s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
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s-> in_sample_fmt, s-> in.ch_count, 0);
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s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
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s->int_sample_fmt, s->out.ch_count, 0);
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s->postin= s->in;
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s->preout= s->out;
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s->midbuf= s->in;
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s->in_buffer= s->in;
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if(!s->resample_first){
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s->midbuf.ch_count= s->out.ch_count;
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s->in_buffer.ch_count = s->out.ch_count;
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}
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s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
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s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
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if(s->rematrix && swr_rematrix_init(s)<0)
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return -1;
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return 0;
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}
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static int realloc_audio(AudioData *a, int count){
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int i, countb;
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AudioData old;
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if(a->count >= count)
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return 0;
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count*=2;
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countb= FFALIGN(count*a->bps, 32);
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old= *a;
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av_assert0(a->planar);
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av_assert0(a->bps);
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av_assert0(a->ch_count);
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a->data= av_malloc(countb*a->ch_count);
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if(!a->data)
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return AVERROR(ENOMEM);
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for(i=0; i<a->ch_count; i++){
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a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
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if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
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}
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av_free(old.data);
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a->count= count;
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return 1;
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}
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static void copy(AudioData *out, AudioData *in,
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int count){
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av_assert0(out->planar == in->planar);
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av_assert0(out->bps == in->bps);
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av_assert0(out->ch_count == in->ch_count);
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if(out->planar){
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int ch;
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for(ch=0; ch<out->ch_count; ch++)
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memcpy(out->ch[ch], in->ch[ch], count*out->bps);
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}else
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memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
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}
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static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
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int i;
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if(out->planar){
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for(i=0; i<out->ch_count; i++)
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out->ch[i]= in_arg[i];
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}else{
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for(i=0; i<out->ch_count; i++)
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out->ch[i]= in_arg[0] + i*out->bps;
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}
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}
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int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
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const uint8_t *in_arg [SWR_CH_MAX], int in_count){
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AudioData *postin, *midbuf, *preout;
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int ret/*, in_max*/;
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AudioData * in= &s->in;
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AudioData *out= &s->out;
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AudioData preout_tmp, midbuf_tmp;
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if(!s->resample){
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if(in_count > out_count)
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return -1;
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out_count = in_count;
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}
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fill_audiodata(in , in_arg);
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fill_audiodata(out, out_arg);
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if(s->full_convert){
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av_assert0(!s->resample);
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swr_audio_convert(s->full_convert, out, in, in_count);
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return out_count;
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}
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// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
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// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
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if((ret=realloc_audio(&s->postin, in_count))<0)
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return ret;
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if(s->resample_first){
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av_assert0(s->midbuf.ch_count == s-> in.ch_count);
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if((ret=realloc_audio(&s->midbuf, out_count))<0)
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return ret;
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}else{
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av_assert0(s->midbuf.ch_count == s->out.ch_count);
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if((ret=realloc_audio(&s->midbuf, in_count))<0)
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return ret;
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}
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if((ret=realloc_audio(&s->preout, out_count))<0)
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return ret;
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postin= &s->postin;
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midbuf_tmp= s->midbuf;
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midbuf= &midbuf_tmp;
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preout_tmp= s->preout;
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preout= &preout_tmp;
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if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
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postin= in;
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if(s->resample_first ? !s->resample : !s->rematrix)
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midbuf= postin;
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if(s->resample_first ? !s->rematrix : !s->resample)
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preout= midbuf;
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if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
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if(preout==in){
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out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
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av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
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copy(out, in, out_count);
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return out_count;
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}
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else if(preout==postin) preout= midbuf= postin= out;
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else if(preout==midbuf) preout= midbuf= out;
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else preout= out;
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}
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if(in != postin){
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swr_audio_convert(s->in_convert, postin, in, in_count);
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}
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if(s->resample_first){
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if(postin != midbuf)
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out_count= resample(s, midbuf, out_count, postin, in_count);
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if(midbuf != preout)
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swr_rematrix(s, preout, midbuf, out_count, preout==out);
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}else{
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if(postin != midbuf)
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swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
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if(midbuf != preout)
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out_count= resample(s, preout, out_count, midbuf, in_count);
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}
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if(preout != out){
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//FIXME packed doesnt need more than 1 chan here!
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swr_audio_convert(s->out_convert, out, preout, out_count);
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}
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return out_count;
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}
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/**
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*
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* out may be equal in.
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*/
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static void buf_set(AudioData *out, AudioData *in, int count){
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if(in->planar){
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int ch;
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for(ch=0; ch<out->ch_count; ch++)
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out->ch[ch]= in->ch[ch] + count*out->bps;
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}else
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out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
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}
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/**
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*
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* @return number of samples output per channel
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*/
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static int resample(SwrContext *s, AudioData *out_param, int out_count,
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const AudioData * in_param, int in_count){
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AudioData in, out, tmp;
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int ret_sum=0;
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int border=0;
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tmp=out=*out_param;
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in = *in_param;
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do{
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int ret, size, consumed;
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if(!s->resample_in_constraint && s->in_buffer_count){
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buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
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ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
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out_count -= ret;
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ret_sum += ret;
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buf_set(&out, &out, ret);
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s->in_buffer_count -= consumed;
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s->in_buffer_index += consumed;
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if(!in_count)
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break;
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if(s->in_buffer_count <= border){
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buf_set(&in, &in, -s->in_buffer_count);
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in_count += s->in_buffer_count;
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s->in_buffer_count=0;
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s->in_buffer_index=0;
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border = 0;
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}
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}
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if(in_count && !s->in_buffer_count){
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s->in_buffer_index=0;
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ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
|
|
out_count -= ret;
|
|
ret_sum += ret;
|
|
buf_set(&out, &out, ret);
|
|
in_count -= consumed;
|
|
buf_set(&in, &in, consumed);
|
|
}
|
|
|
|
//TODO is this check sane considering the advanced copy avoidance below
|
|
size= s->in_buffer_index + s->in_buffer_count + in_count;
|
|
if( size > s->in_buffer.count
|
|
&& s->in_buffer_count + in_count <= s->in_buffer_index){
|
|
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
|
|
copy(&s->in_buffer, &tmp, s->in_buffer_count);
|
|
s->in_buffer_index=0;
|
|
}else
|
|
if((ret=realloc_audio(&s->in_buffer, size)) < 0)
|
|
return ret;
|
|
|
|
if(in_count){
|
|
int count= in_count;
|
|
if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
|
|
|
|
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
|
|
copy(&tmp, &in, /*in_*/count);
|
|
s->in_buffer_count += count;
|
|
in_count -= count;
|
|
border += count;
|
|
buf_set(&in, &in, count);
|
|
s->resample_in_constraint= 0;
|
|
if(s->in_buffer_count != count || in_count)
|
|
continue;
|
|
}
|
|
break;
|
|
}while(1);
|
|
|
|
s->resample_in_constraint= !!out_count;
|
|
|
|
return ret_sum;
|
|
}
|