/* * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) * * This file is part of libswresample * * libswresample is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * libswresample is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with libswresample; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/opt.h" #include "swresample_internal.h" #include "audioconvert.h" #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include #define C30DB M_SQRT2 #define C15DB 1.189207115 #define C__0DB 1.0 #define C_15DB 0.840896415 #define C_30DB M_SQRT1_2 #define C_45DB 0.594603558 #define C_60DB 0.5 #define ALIGN 32 //TODO split options array out? #define OFFSET(x) offsetof(SwrContext,x) #define PARAM AV_OPT_FLAG_AUDIO_PARAM static const AVOption options[]={ {"ich" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"in_channel_count" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"och" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"out_channel_count" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"uch" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"used_channel_count" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, {"isr" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, {"in_sample_rate" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, {"osr" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, {"out_sample_rate" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, {"isf" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"in_sample_fmt" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"osf" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"out_sample_fmt" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"tsf" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"internal_sample_fmt" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, {"icl" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, {"in_channel_layout" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, {"ocl" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, {"out_channel_layout" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, {"clev" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, {"center_mix_level" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, {"slev" , "set surround mix level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, {"surround_mix_level" , "set surround mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, {"lfe_mix_level" , "set LFE mix level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM}, {"rmvol" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, {"rematrix_volume" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, {"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"}, {"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"}, {"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"}, {"dither_scale" , "set dither scale" , OFFSET(dither.scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM}, {"dither_method" , "set dither method" , OFFSET(dither.method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"}, {"rectangular" , "select rectangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"}, {"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"}, {"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"lipshitz" , "select lipshitz noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"shibata" , "select shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"low_shibata" , "select low shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"high_shibata" , "select high shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"f_weighted" , "select f-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"modified_e_weighted" , "select modified-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"improved_e_weighted" , "select improved-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, {"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM }, {"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM }, {"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, {"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM }, /* duplicate option in order to work with avconv */ {"resample_cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM }, {"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"}, {"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"}, {"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"}, {"precision" , "set soxr resampling precision (in bits)" , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM }, {"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation" , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, {"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied" , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM }, {"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data." , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM }, {"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps." , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM }, {"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps." , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM }, {"async" , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)" , OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM }, {"first_pts" , "Assume the first pts should be this value (in samples)." , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE }, INT64_MIN,INT64_MAX, PARAM }, { "matrix_encoding" , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" }, { "none", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, { "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, { "filter_type" , "select swr 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" }, { "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" }, { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" }, { "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" }, { "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM }, { "output_sample_bits" , "" , OFFSET(dither.output_sample_bits) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , 0 }, {0} }; static const char* context_to_name(void* ptr) { return "SWR"; } static const AVClass av_class = { .class_name = "SWResampler", .item_name = context_to_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .log_level_offset_offset = OFFSET(log_level_offset), .parent_log_context_offset = OFFSET(log_ctx), .category = AV_CLASS_CATEGORY_SWRESAMPLER, }; unsigned swresample_version(void) { av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100); return LIBSWRESAMPLE_VERSION_INT; } const char *swresample_configuration(void) { return FFMPEG_CONFIGURATION; } const char *swresample_license(void) { #define LICENSE_PREFIX "libswresample license: " return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; } int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){ if(!s || s->in_convert) // s needs to be allocated but not initialized return AVERROR(EINVAL); s->channel_map = channel_map; return 0; } const AVClass *swr_get_class(void) { return &av_class; } av_cold struct SwrContext *swr_alloc(void){ SwrContext *s= av_mallocz(sizeof(SwrContext)); if(s){ s->av_class= &av_class; av_opt_set_defaults(s); } return s; } struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx){ if(!s) s= swr_alloc(); if(!s) return NULL; s->log_level_offset= log_offset; s->log_ctx= log_ctx; av_opt_set_int(s, "ocl", out_ch_layout, 0); av_opt_set_int(s, "osf", out_sample_fmt, 0); av_opt_set_int(s, "osr", out_sample_rate, 0); av_opt_set_int(s, "icl", in_ch_layout, 0); av_opt_set_int(s, "isf", in_sample_fmt, 0); av_opt_set_int(s, "isr", in_sample_rate, 0); av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0); av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0); av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0); av_opt_set_int(s, "uch", 0, 0); return s; } static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){ a->fmt = fmt; a->bps = av_get_bytes_per_sample(fmt); a->planar= av_sample_fmt_is_planar(fmt); } static void free_temp(AudioData *a){ av_free(a->data); memset(a, 0, sizeof(*a)); } av_cold void swr_free(SwrContext **ss){ SwrContext *s= *ss; if(s){ free_temp(&s->postin); free_temp(&s->midbuf); free_temp(&s->preout); free_temp(&s->in_buffer); free_temp(&s->silence); free_temp(&s->drop_temp); free_temp(&s->dither.noise); free_temp(&s->dither.temp); swri_audio_convert_free(&s-> in_convert); swri_audio_convert_free(&s->out_convert); swri_audio_convert_free(&s->full_convert); if (s->resampler) s->resampler->free(&s->resample); swri_rematrix_free(s); } av_freep(ss); } av_cold int swr_init(struct SwrContext *s){ int ret; s->in_buffer_index= 0; s->in_buffer_count= 0; s->resample_in_constraint= 0; free_temp(&s->postin); free_temp(&s->midbuf); free_temp(&s->preout); free_temp(&s->in_buffer); free_temp(&s->silence); free_temp(&s->drop_temp); free_temp(&s->dither.noise); free_temp(&s->dither.temp); memset(s->in.ch, 0, sizeof(s->in.ch)); memset(s->out.ch, 0, sizeof(s->out.ch)); swri_audio_convert_free(&s-> in_convert); swri_audio_convert_free(&s->out_convert); swri_audio_convert_free(&s->full_convert); swri_rematrix_free(s); s->flushed = 0; if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){ av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt); return AVERROR(EINVAL); } if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){ av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt); return AVERROR(EINVAL); } if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) { av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout); s->in_ch_layout = 0; } if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) { av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout); s->out_ch_layout = 0; } switch(s->engine){ #if CONFIG_LIBSOXR extern struct Resampler const soxr_resampler; case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break; #endif case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; default: av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); return AVERROR(EINVAL); } if(!s->used_ch_count) s->used_ch_count= s->in.ch_count; if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){ av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n"); s-> in_ch_layout= 0; } if(!s-> in_ch_layout) s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count); if(!s->out_ch_layout) s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count); s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 || s->rematrix_custom; if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){ if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){ s->int_sample_fmt= AV_SAMPLE_FMT_S16P; }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P && !s->rematrix && s->engine != SWR_ENGINE_SOXR){ s->int_sample_fmt= AV_SAMPLE_FMT_S32P; }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){ s->int_sample_fmt= AV_SAMPLE_FMT_FLTP; }else{ av_log(s, AV_LOG_DEBUG, "Using double precision mode\n"); s->int_sample_fmt= AV_SAMPLE_FMT_DBLP; } } if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ 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)); return AVERROR(EINVAL); } set_audiodata_fmt(&s-> in, s-> in_sample_fmt); set_audiodata_fmt(&s->out, s->out_sample_fmt); if (s->firstpts_in_samples != AV_NOPTS_VALUE) { if (!s->async && s->min_compensation >= FLT_MAX/2) s->async = 1; s->firstpts = s->outpts = s->firstpts_in_samples * s->out_sample_rate; } else s->firstpts = AV_NOPTS_VALUE; if (s->async) { if (s->min_compensation >= FLT_MAX/2) s->min_compensation = 0.001; if (s->async > 1.0001) { s->max_soft_compensation = s->async / (double) s->in_sample_rate; } } if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ s->resample = s->resampler->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, s->precision, s->cheby); }else s->resampler->free(&s->resample); if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P && s->int_sample_fmt != AV_SAMPLE_FMT_S32P && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP && s->resample){ av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n"); return -1; } #define RSC 1 //FIXME finetune if(!s-> in.ch_count) s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout); if(!s->used_ch_count) s->used_ch_count= s->in.ch_count; if(!s->out.ch_count) s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout); if(!s-> in.ch_count){ av_assert0(!s->in_ch_layout); av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n"); return -1; } if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) { char l1[1024], l2[1024]; av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout); av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout); av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s " "but there is not enough information to do it\n", l1, l2); return -1; } av_assert0(s->used_ch_count); av_assert0(s->out.ch_count); s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0; s->in_buffer= s->in; s->silence = s->in; s->drop_temp= s->out; if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){ s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt, s-> in_sample_fmt, s-> in.ch_count, NULL, 0); return 0; } s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt, s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0); s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt, s->int_sample_fmt, s->out.ch_count, NULL, 0); if (!s->in_convert || !s->out_convert) return AVERROR(ENOMEM); s->postin= s->in; s->preout= s->out; s->midbuf= s->in; if(s->channel_map){ s->postin.ch_count= s->midbuf.ch_count= s->used_ch_count; if(s->resample) s->in_buffer.ch_count= s->used_ch_count; } if(!s->resample_first){ 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); } if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) return ret; if(s->rematrix || s->dither.method) return swri_rematrix_init(s); return 0; } int swri_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; ich_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; chch_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; ich_count; i++) out->ch[i]= in_arg[i]; }else{ for(i=0; ich_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; ich_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; chch_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= s->resampler->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((s->flushed || in_count) && !s->in_buffer_count){ s->in_buffer_index=0; ret= s->resampler->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=swri_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=swri_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=swri_realloc_audio(&s->midbuf, out_count))<0) return ret; }else{ av_assert0(s->midbuf.ch_count == s->out.ch_count); if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) return ret; } if((ret=swri_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){ AudioData *conv_src = preout; if(s->dither.method){ int ch; int dither_count= FFMAX(out_count, 1<<16); if (preout == in) { conv_src = &s->dither.temp; if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) return ret; } if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) return ret; if(ret) for(ch=0; chdither.noise.ch_count; ch++) swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<dither.noise.fmt); av_assert0(s->dither.noise.ch_count == preout->ch_count); if(s->dither.noise_pos + out_count > s->dither.noise.count) s->dither.noise_pos = 0; if (s->dither.method < SWR_DITHER_NS){ if (s->mix_2_1_simd) { int len1= out_count&~15; int off = len1 * preout->bps; if(len1) for(ch=0; chch_count; ch++) s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, len1); if(out_count != len1) for(ch=0; chch_count; ch++) s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1); } else { for(ch=0; chch_count; ch++) s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count); } } else { switch(s->int_sample_fmt) { case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; } } s->dither.noise_pos += out_count; } //FIXME packed doesnt need more than 1 chan here! swri_audio_convert(s->out_convert, out, conv_src, 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; while(s->drop_output > 0){ int ret; uint8_t *tmp_arg[SWR_CH_MAX]; #define MAX_DROP_STEP 16384 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) return ret; reversefill_audiodata(&s->drop_temp, tmp_arg); s->drop_output *= -1; //FIXME find a less hackish solution ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter s->drop_output *= -1; in_count = 0; if(ret>0) { s->drop_output -= ret; continue; } if(s->drop_output || !out_arg) return 0; } if(!in_arg){ if(s->resample){ if (!s->flushed) s->resampler->flush(s); s->resample_in_constraint = 0; s->flushed = 1; }else if(!s->in_buffer_count){ 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=swri_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; uint8_t *tmp_arg[SWR_CH_MAX]; if(count <= 0) return 0; #define MAX_SILENCE_STEP 16384 while (count > MAX_SILENCE_STEP) { if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) return ret; count -= MAX_SILENCE_STEP; } if((ret=swri_realloc_audio(&s->silence, count))<0) return ret; if(s->silence.planar) for(i=0; isilence.ch_count; i++) { memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); } else memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); reversefill_audiodata(&s->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); return ret; } int64_t swr_get_delay(struct SwrContext *s, int64_t base){ if (s->resampler && s->resample){ return s->resampler->get_delay(s, base); }else{ return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; } } int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ int ret; if (!s || compensation_distance < 0) return AVERROR(EINVAL); if (!compensation_distance && sample_delta) return AVERROR(EINVAL); if (!s->resample) { s->flags |= SWR_FLAG_RESAMPLE; ret = swr_init(s); if (ret < 0) return ret; } if (!s->resampler->set_compensation){ return AVERROR(EINVAL); }else{ return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); } } int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ if(pts == INT64_MIN) return s->outpts; if (s->firstpts == AV_NOPTS_VALUE) s->outpts = s->firstpts = pts; 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 + s->drop_output*(int64_t)s->in_sample_rate; double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); if(fabs(fdelta) > s->min_compensation) { if(s->outpts == s->firstpts || 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; } }