diff --git a/configure b/configure index 7cf8d9d210..7f3202e14f 100755 --- a/configure +++ b/configure @@ -213,6 +213,7 @@ External library support: --enable-libpulse enable Pulseaudio input via libpulse [no] --enable-librtmp enable RTMP[E] support via librtmp [no] --enable-libschroedinger enable Dirac de/encoding via libschroedinger [no] + --enable-libsoxr enable Include libsoxr resampling [no] --enable-libspeex enable Speex de/encoding via libspeex [no] --enable-libstagefright-h264 enable H.264 decoding via libstagefright [no] --enable-libtheora enable Theora encoding via libtheora [no] @@ -1173,6 +1174,7 @@ CONFIG_LIST=" libpulse librtmp libschroedinger + libsoxr libspeex libstagefright_h264 libtheora @@ -3839,6 +3841,7 @@ enabled libopus && require_pkg_config opus opus_multistream.h opus_multistrea enabled libpulse && require_pkg_config libpulse-simple pulse/simple.h pa_simple_new enabled librtmp && require_pkg_config librtmp librtmp/rtmp.h RTMP_Socket enabled libschroedinger && require_pkg_config schroedinger-1.0 schroedinger/schro.h schro_init +enabled libsoxr && require libsoxr soxr.h soxr_create -lsoxr enabled libspeex && require libspeex speex/speex.h speex_decoder_init -lspeex enabled libstagefright_h264 && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h media/stagefright/MediaBufferGroup.h media/stagefright/MediaDebug.h media/stagefright/MediaDefs.h @@ -4254,6 +4257,7 @@ echo "libopus enabled ${libopus-no}" echo "libpulse enabled ${libpulse-no}" echo "librtmp enabled ${librtmp-no}" echo "libschroedinger enabled ${libschroedinger-no}" +echo "libsoxr enabled ${libsoxr-no}" echo "libspeex enabled ${libspeex-no}" echo "libstagefright-h264 enabled ${libstagefright_h264-no}" echo "libtheora enabled ${libtheora-no}" diff --git a/libswresample/Makefile b/libswresample/Makefile index 4c3ec1cff5..9d9f10c6c6 100644 --- a/libswresample/Makefile +++ b/libswresample/Makefile @@ -13,4 +13,6 @@ OBJS = audioconvert.o \ resample.o \ swresample.o \ +OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o + TESTPROGS = swresample diff --git a/libswresample/resample.c b/libswresample/resample.c index 7256fcf41a..60ba0e9bd0 100644 --- a/libswresample/resample.c +++ b/libswresample/resample.c @@ -196,7 +196,8 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap } static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, - double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta){ + double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, + double precision, int cheby){ double cutoff = cutoff0? cutoff0 : 0.8; double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); int phase_count= 1< + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * audio resampling with soxr + */ + +#include "libavutil/log.h" +#include "swresample_internal.h" + +#include + +static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, + double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){ + soxr_error_t error; + + soxr_datatype_t type = + format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S : + format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I : + format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S : + format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I : + format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S : + format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I : + format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S : + format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1; + + soxr_io_spec_t io_spec = soxr_io_spec(type, type); + + soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby); + q_spec.bits = linear? 0 : precision; + q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc; + + soxr_delete((soxr_t)c); + c = (struct ResampleContext *) + soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0); + if (!c) + av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error); + return c; +} + +static void destroy(struct ResampleContext * *c){ + soxr_delete((soxr_t)*c); + *c = NULL; +} + +static int flush(struct SwrContext *s){ + soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL); + return 0; +} + +static int process( + struct ResampleContext * c, AudioData *dst, int dst_size, + AudioData *src, int src_size, int *consumed){ + size_t idone, odone; + soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count)); + error = soxr_process((soxr_t)c, src->ch, (size_t)src_size, + &idone, dst->ch, (size_t)dst_size, &odone); + *consumed = (int)idone; + return error? -1 : odone; +} + +static int64_t get_delay(struct SwrContext *s, int64_t base){ + double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate; + return (int64_t)(delay_s * base + .5); +} + +struct Resampler const soxr_resampler={ + create, destroy, process, flush, NULL /* set_compensation */, get_delay, +}; + diff --git a/libswresample/swresample.c b/libswresample/swresample.c index 207abb4e3e..af983a1b4e 100644 --- a/libswresample/swresample.c +++ b/libswresample/swresample.c @@ -86,6 +86,9 @@ static const AVOption options[]={ {"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 resampling precision" , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM }, +{"cheby" , "enable Chebyshev passband" , 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." @@ -262,6 +265,10 @@ av_cold int swr_init(struct SwrContext *s){ } 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"); @@ -272,7 +279,7 @@ av_cold int swr_init(struct SwrContext *s){ set_audiodata_fmt(&s->out, s->out_sample_fmt); 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->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 @@ -491,7 +498,7 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count, } } - if(in_count && !s->in_buffer_count){ + 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; diff --git a/libswresample/swresample.h b/libswresample/swresample.h index 356fb61488..a2b5cd0f09 100644 --- a/libswresample/swresample.h +++ b/libswresample/swresample.h @@ -117,6 +117,7 @@ enum SwrDitherType { /** Resampling Engines */ enum SwrEngine { SWR_ENGINE_SWR, /**< SW Resampler */ + SWR_ENGINE_SOXR, /**< SoX Resampler */ SWR_ENGINE_NB, ///< not part of API/ABI }; diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h index bc53e6a699..19ca0d0fa3 100644 --- a/libswresample/swresample_internal.h +++ b/libswresample/swresample_internal.h @@ -77,6 +77,8 @@ struct SwrContext { double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */ enum SwrFilterType filter_type; /**< resampling filter type */ int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ + double precision; /**< resampling precision (in bits) */ + int cheby; /**< if 1 then the resampling FIR filter will be configured for maximal passband flatness */ float min_compensation; ///< minimum below which no compensation will happen float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen @@ -125,7 +127,7 @@ struct SwrContext { }; typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, - double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta); + double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby); typedef void (* resample_free_func)(struct ResampleContext **c); typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); typedef int (* resample_flush_func)(struct SwrContext *c);