mirror of https://git.ffmpeg.org/ffmpeg.git
790 lines
33 KiB
C
790 lines
33 KiB
C
/*
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* Copyright (C) 2011-2012 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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#include <float.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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#define ALIGN 32
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//TODO split options array out?
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#define OFFSET(x) offsetof(SwrContext,x)
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#define PARAM AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption options[]={
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{"ich" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
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{"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
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{"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
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{"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
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{"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
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{"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
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{"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
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{"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
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{"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
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{"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
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{"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
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{"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
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{"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
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{"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
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{"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
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{"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
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{"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
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{"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
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{"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
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{"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
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{"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
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{"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
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{"swr_flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
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{"res" , "Force Resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
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{"dither_scale" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
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{"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
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{"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
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{"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
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{"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
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{"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
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{"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
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{"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
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{"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
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{"min_comp" , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
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, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
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{"min_hard_comp" , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
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, OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
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{"comp_duration" , "Duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
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, OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
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{"max_soft_comp" , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
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, OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
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{ "filter_type" , "Filter Type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
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{ "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
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{ "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
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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 = {
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.class_name = "SWResampler",
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.item_name = context_to_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.log_level_offset_offset = OFFSET(log_level_offset),
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.parent_log_context_offset = OFFSET(log_ctx),
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.category = AV_CLASS_CATEGORY_SWRESAMPLER,
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};
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unsigned swresample_version(void)
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{
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av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
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return LIBSWRESAMPLE_VERSION_INT;
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}
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const char *swresample_configuration(void)
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{
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return FFMPEG_CONFIGURATION;
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}
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const char *swresample_license(void)
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{
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#define LICENSE_PREFIX "libswresample license: "
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return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
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}
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int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
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if(!s || s->in_convert) // s needs to be allocated but not initialized
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return AVERROR(EINVAL);
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s->channel_map = channel_map;
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return 0;
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}
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const AVClass *swr_get_class(void)
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{
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return &av_class;
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}
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struct 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_defaults(s);
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}
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return s;
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}
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struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
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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_opt_set_int(s, "ocl", out_ch_layout, 0);
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av_opt_set_int(s, "osf", out_sample_fmt, 0);
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av_opt_set_int(s, "osr", out_sample_rate, 0);
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av_opt_set_int(s, "icl", in_ch_layout, 0);
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av_opt_set_int(s, "isf", in_sample_fmt, 0);
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av_opt_set_int(s, "isr", in_sample_rate, 0);
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av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
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av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
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av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
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av_opt_set_int(s, "uch", 0, 0);
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return s;
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}
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static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
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a->fmt = fmt;
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a->bps = av_get_bytes_per_sample(fmt);
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a->planar= av_sample_fmt_is_planar(fmt);
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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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free_temp(&s->dither);
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swri_audio_convert_free(&s-> in_convert);
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swri_audio_convert_free(&s->out_convert);
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swri_audio_convert_free(&s->full_convert);
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swri_resample_free(&s->resample);
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swri_rematrix_free(s);
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}
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av_freep(ss);
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}
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int swr_init(struct 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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free_temp(&s->dither);
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swri_audio_convert_free(&s-> in_convert);
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swri_audio_convert_free(&s->out_convert);
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swri_audio_convert_free(&s->full_convert);
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swri_rematrix_free(s);
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s->flushed = 0;
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if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
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av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", 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 output sample format %d is invalid\n", 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_NONE){
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if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
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s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
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}else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
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s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
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}else{
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av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
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s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
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}
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}
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
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&&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
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&&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
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&&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
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av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL 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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set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
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set_audiodata_fmt(&s->out, s->out_sample_fmt);
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if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
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s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
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}else
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swri_resample_free(&s->resample);
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if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_S32P
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&& s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
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&& s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
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&& s->resample){
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av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
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return -1;
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}
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if(!s->used_ch_count)
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s->used_ch_count= s->in.ch_count;
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if(s->used_ch_count && s-> in_ch_layout && s->used_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 the number of used channels, 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->used_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 || s->rematrix_volume!=1.0 ||
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s->rematrix_custom;
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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->used_ch_count)
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s->used_ch_count= s->in.ch_count;
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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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if(!s-> in.ch_count){
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av_assert0(!s->in_ch_layout);
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av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
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return -1;
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}
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if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
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av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
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return -1;
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}
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av_assert0(s->used_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_buffer= s->in;
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if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
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s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
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s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
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return 0;
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}
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s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
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s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
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s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
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s->int_sample_fmt, s->out.ch_count, NULL, 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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if(s->channel_map){
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s->postin.ch_count=
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s->midbuf.ch_count= s->used_ch_count;
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if(s->resample)
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s->in_buffer.ch_count= s->used_ch_count;
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}
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if(!s->resample_first){
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s->midbuf.ch_count= s->out.ch_count;
|
|
if(s->resample)
|
|
s->in_buffer.ch_count = s->out.ch_count;
|
|
}
|
|
|
|
set_audiodata_fmt(&s->postin, s->int_sample_fmt);
|
|
set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
|
|
set_audiodata_fmt(&s->preout, s->int_sample_fmt);
|
|
|
|
if(s->resample){
|
|
set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
|
|
}
|
|
|
|
s->dither = s->preout;
|
|
|
|
if(s->rematrix || s->dither_method)
|
|
return swri_rematrix_init(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int realloc_audio(AudioData *a, int count){
|
|
int i, countb;
|
|
AudioData old;
|
|
|
|
if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
|
|
return AVERROR(EINVAL);
|
|
|
|
if(a->count >= count)
|
|
return 0;
|
|
|
|
count*=2;
|
|
|
|
countb= FFALIGN(count*a->bps, ALIGN);
|
|
old= *a;
|
|
|
|
av_assert0(a->bps);
|
|
av_assert0(a->ch_count);
|
|
|
|
a->data= av_mallocz(countb*a->ch_count);
|
|
if(!a->data)
|
|
return AVERROR(ENOMEM);
|
|
for(i=0; i<a->ch_count; i++){
|
|
a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
|
|
if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
|
|
}
|
|
if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
|
|
av_free(old.data);
|
|
a->count= count;
|
|
|
|
return 1;
|
|
}
|
|
|
|
static void copy(AudioData *out, AudioData *in,
|
|
int count){
|
|
av_assert0(out->planar == in->planar);
|
|
av_assert0(out->bps == in->bps);
|
|
av_assert0(out->ch_count == in->ch_count);
|
|
if(out->planar){
|
|
int ch;
|
|
for(ch=0; ch<out->ch_count; ch++)
|
|
memcpy(out->ch[ch], in->ch[ch], count*out->bps);
|
|
}else
|
|
memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
|
|
}
|
|
|
|
static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
|
|
int i;
|
|
if(!in_arg){
|
|
memset(out->ch, 0, sizeof(out->ch));
|
|
}else if(out->planar){
|
|
for(i=0; i<out->ch_count; i++)
|
|
out->ch[i]= in_arg[i];
|
|
}else{
|
|
for(i=0; i<out->ch_count; i++)
|
|
out->ch[i]= in_arg[0] + i*out->bps;
|
|
}
|
|
}
|
|
|
|
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
|
|
int i;
|
|
if(out->planar){
|
|
for(i=0; i<out->ch_count; i++)
|
|
in_arg[i]= out->ch[i];
|
|
}else{
|
|
in_arg[0]= out->ch[0];
|
|
}
|
|
}
|
|
|
|
/**
|
|
*
|
|
* out may be equal in.
|
|
*/
|
|
static void buf_set(AudioData *out, AudioData *in, int count){
|
|
int ch;
|
|
if(in->planar){
|
|
for(ch=0; ch<out->ch_count; ch++)
|
|
out->ch[ch]= in->ch[ch] + count*out->bps;
|
|
}else{
|
|
for(ch=out->ch_count-1; ch>=0; ch--)
|
|
out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
|
|
}
|
|
}
|
|
|
|
/**
|
|
*
|
|
* @return number of samples output per channel
|
|
*/
|
|
static int resample(SwrContext *s, AudioData *out_param, int out_count,
|
|
const AudioData * in_param, int in_count){
|
|
AudioData in, out, tmp;
|
|
int ret_sum=0;
|
|
int border=0;
|
|
|
|
av_assert1(s->in_buffer.ch_count == in_param->ch_count);
|
|
av_assert1(s->in_buffer.planar == in_param->planar);
|
|
av_assert1(s->in_buffer.fmt == in_param->fmt);
|
|
|
|
tmp=out=*out_param;
|
|
in = *in_param;
|
|
|
|
do{
|
|
int ret, size, consumed;
|
|
if(!s->resample_in_constraint && s->in_buffer_count){
|
|
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
|
|
ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
|
|
out_count -= ret;
|
|
ret_sum += ret;
|
|
buf_set(&out, &out, ret);
|
|
s->in_buffer_count -= consumed;
|
|
s->in_buffer_index += consumed;
|
|
|
|
if(!in_count)
|
|
break;
|
|
if(s->in_buffer_count <= border){
|
|
buf_set(&in, &in, -s->in_buffer_count);
|
|
in_count += s->in_buffer_count;
|
|
s->in_buffer_count=0;
|
|
s->in_buffer_index=0;
|
|
border = 0;
|
|
}
|
|
}
|
|
|
|
if(in_count && !s->in_buffer_count){
|
|
s->in_buffer_index=0;
|
|
ret= swri_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;
|
|
}
|
|
|
|
static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
|
|
AudioData *in , int in_count){
|
|
AudioData *postin, *midbuf, *preout;
|
|
int ret/*, in_max*/;
|
|
AudioData preout_tmp, midbuf_tmp;
|
|
|
|
if(s->full_convert){
|
|
av_assert0(!s->resample);
|
|
swri_audio_convert(s->full_convert, out, in, in_count);
|
|
return out_count;
|
|
}
|
|
|
|
// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
|
|
// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
|
|
|
|
if((ret=realloc_audio(&s->postin, in_count))<0)
|
|
return ret;
|
|
if(s->resample_first){
|
|
av_assert0(s->midbuf.ch_count == s->used_ch_count);
|
|
if((ret=realloc_audio(&s->midbuf, out_count))<0)
|
|
return ret;
|
|
}else{
|
|
av_assert0(s->midbuf.ch_count == s->out.ch_count);
|
|
if((ret=realloc_audio(&s->midbuf, in_count))<0)
|
|
return ret;
|
|
}
|
|
if((ret=realloc_audio(&s->preout, out_count))<0)
|
|
return ret;
|
|
|
|
postin= &s->postin;
|
|
|
|
midbuf_tmp= s->midbuf;
|
|
midbuf= &midbuf_tmp;
|
|
preout_tmp= s->preout;
|
|
preout= &preout_tmp;
|
|
|
|
if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
|
|
postin= in;
|
|
|
|
if(s->resample_first ? !s->resample : !s->rematrix)
|
|
midbuf= postin;
|
|
|
|
if(s->resample_first ? !s->rematrix : !s->resample)
|
|
preout= midbuf;
|
|
|
|
if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
|
|
if(preout==in){
|
|
out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
|
|
av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
|
|
copy(out, in, out_count);
|
|
return out_count;
|
|
}
|
|
else if(preout==postin) preout= midbuf= postin= out;
|
|
else if(preout==midbuf) preout= midbuf= out;
|
|
else preout= out;
|
|
}
|
|
|
|
if(in != postin){
|
|
swri_audio_convert(s->in_convert, postin, in, in_count);
|
|
}
|
|
|
|
if(s->resample_first){
|
|
if(postin != midbuf)
|
|
out_count= resample(s, midbuf, out_count, postin, in_count);
|
|
if(midbuf != preout)
|
|
swri_rematrix(s, preout, midbuf, out_count, preout==out);
|
|
}else{
|
|
if(postin != midbuf)
|
|
swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
|
|
if(midbuf != preout)
|
|
out_count= resample(s, preout, out_count, midbuf, in_count);
|
|
}
|
|
|
|
if(preout != out && out_count){
|
|
if(s->dither_method){
|
|
int ch;
|
|
int dither_count= FFMAX(out_count, 1<<16);
|
|
av_assert0(preout != in);
|
|
|
|
if((ret=realloc_audio(&s->dither, dither_count))<0)
|
|
return ret;
|
|
if(ret)
|
|
for(ch=0; ch<s->dither.ch_count; ch++)
|
|
swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
|
|
av_assert0(s->dither.ch_count == preout->ch_count);
|
|
|
|
if(s->dither_pos + out_count > s->dither.count)
|
|
s->dither_pos = 0;
|
|
|
|
for(ch=0; ch<preout->ch_count; ch++)
|
|
s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
|
|
|
|
s->dither_pos += out_count;
|
|
}
|
|
//FIXME packed doesnt need more than 1 chan here!
|
|
swri_audio_convert(s->out_convert, out, preout, out_count);
|
|
}
|
|
return out_count;
|
|
}
|
|
|
|
int 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){
|
|
AudioData * in= &s->in;
|
|
AudioData *out= &s->out;
|
|
|
|
if(s->drop_output > 0){
|
|
int ret;
|
|
AudioData tmp = s->out;
|
|
uint8_t *tmp_arg[SWR_CH_MAX];
|
|
tmp.count = 0;
|
|
tmp.data = NULL;
|
|
if((ret=realloc_audio(&tmp, s->drop_output))<0)
|
|
return ret;
|
|
|
|
reversefill_audiodata(&tmp, tmp_arg);
|
|
s->drop_output *= -1; //FIXME find a less hackish solution
|
|
ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
|
|
s->drop_output *= -1;
|
|
if(ret>0)
|
|
s->drop_output -= ret;
|
|
|
|
av_freep(&tmp.data);
|
|
if(s->drop_output || !out_arg)
|
|
return 0;
|
|
in_count = 0;
|
|
}
|
|
|
|
if(!in_arg){
|
|
if(s->in_buffer_count){
|
|
if (s->resample && !s->flushed) {
|
|
AudioData *a= &s->in_buffer;
|
|
int i, j, ret;
|
|
if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
|
|
return ret;
|
|
av_assert0(a->planar);
|
|
for(i=0; i<a->ch_count; i++){
|
|
for(j=0; j<s->in_buffer_count; j++){
|
|
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
|
|
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
|
|
}
|
|
}
|
|
s->in_buffer_count += (s->in_buffer_count+1)/2;
|
|
s->resample_in_constraint = 0;
|
|
s->flushed = 1;
|
|
}
|
|
}else{
|
|
return 0;
|
|
}
|
|
}else
|
|
fill_audiodata(in , (void*)in_arg);
|
|
|
|
fill_audiodata(out, out_arg);
|
|
|
|
if(s->resample){
|
|
int ret = swr_convert_internal(s, out, out_count, in, in_count);
|
|
if(ret>0 && !s->drop_output)
|
|
s->outpts += ret * (int64_t)s->in_sample_rate;
|
|
return ret;
|
|
}else{
|
|
AudioData tmp= *in;
|
|
int ret2=0;
|
|
int ret, size;
|
|
size = FFMIN(out_count, s->in_buffer_count);
|
|
if(size){
|
|
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
|
|
ret= swr_convert_internal(s, out, size, &tmp, size);
|
|
if(ret<0)
|
|
return ret;
|
|
ret2= ret;
|
|
s->in_buffer_count -= ret;
|
|
s->in_buffer_index += ret;
|
|
buf_set(out, out, ret);
|
|
out_count -= ret;
|
|
if(!s->in_buffer_count)
|
|
s->in_buffer_index = 0;
|
|
}
|
|
|
|
if(in_count){
|
|
size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
|
|
|
|
if(in_count > out_count) { //FIXME move after swr_convert_internal
|
|
if( size > s->in_buffer.count
|
|
&& s->in_buffer_count + in_count - out_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(out_count){
|
|
size = FFMIN(in_count, out_count);
|
|
ret= swr_convert_internal(s, out, size, in, size);
|
|
if(ret<0)
|
|
return ret;
|
|
buf_set(in, in, ret);
|
|
in_count -= ret;
|
|
ret2 += ret;
|
|
}
|
|
if(in_count){
|
|
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
|
|
copy(&tmp, in, in_count);
|
|
s->in_buffer_count += in_count;
|
|
}
|
|
}
|
|
if(ret2>0 && !s->drop_output)
|
|
s->outpts += ret2 * (int64_t)s->in_sample_rate;
|
|
return ret2;
|
|
}
|
|
}
|
|
|
|
int swr_drop_output(struct SwrContext *s, int count){
|
|
s->drop_output += count;
|
|
|
|
if(s->drop_output <= 0)
|
|
return 0;
|
|
|
|
av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
|
|
return swr_convert(s, NULL, s->drop_output, NULL, 0);
|
|
}
|
|
|
|
int swr_inject_silence(struct SwrContext *s, int count){
|
|
int ret, i;
|
|
AudioData silence = s->in;
|
|
uint8_t *tmp_arg[SWR_CH_MAX];
|
|
|
|
if(count <= 0)
|
|
return 0;
|
|
|
|
silence.count = 0;
|
|
silence.data = NULL;
|
|
if((ret=realloc_audio(&silence, count))<0)
|
|
return ret;
|
|
|
|
if(silence.planar) for(i=0; i<silence.ch_count; i++) {
|
|
memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
|
|
} else
|
|
memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
|
|
|
|
reversefill_audiodata(&silence, tmp_arg);
|
|
av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
|
|
ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
|
|
av_freep(&silence.data);
|
|
return ret;
|
|
}
|
|
|
|
int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
|
|
if(pts == INT64_MIN)
|
|
return s->outpts;
|
|
if(s->min_compensation >= FLT_MAX) {
|
|
return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
|
|
} else {
|
|
int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
|
|
double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
|
|
|
|
if(fabs(fdelta) > s->min_compensation) {
|
|
if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
|
|
int ret;
|
|
if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
|
|
else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
|
|
if(ret<0){
|
|
av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
|
|
}
|
|
} else if(s->soft_compensation_duration && s->max_soft_compensation) {
|
|
int duration = s->out_sample_rate * s->soft_compensation_duration;
|
|
double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
|
|
int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
|
|
av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
|
|
swr_set_compensation(s, comp, duration);
|
|
}
|
|
}
|
|
|
|
return s->outpts;
|
|
}
|
|
}
|