| 0,0 → 1,934 |
|---|
| /* |
| * 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 <float.h> |
| |
| #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_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, |
| {"in_channel_layout" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, |
| {"ocl" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, |
| {"out_channel_layout" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.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}, |
| {"rematrix_maxval" , "set rematrix maxval" , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0 }, 0 , 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" , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , PARAM }, |
| {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; 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_freep(&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= 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 |
| && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){ |
| 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; ch<s->dither.noise.ch_count; ch++) |
| swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->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; ch<preout->ch_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_simd_one, 0, 0, len1); |
| if(out_count != len1) |
| for(ch=0; ch<preout->ch_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; ch<preout->ch_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 doesn't 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 doesn't 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; i<s->silence.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; |
| } |
| } |