diff options
Diffstat (limited to 'libswresample/swresample.c')
-rw-r--r-- | libswresample/swresample.c | 795 |
1 files changed, 795 insertions, 0 deletions
diff --git a/libswresample/swresample.c b/libswresample/swresample.c new file mode 100644 index 0000000..5d02aaf --- /dev/null +++ b/libswresample/swresample.c @@ -0,0 +1,795 @@ +/* + * Copyright (C) 2011-2012 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/audioconvert.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" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM}, +{"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM}, +{"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM}, +{"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM}, +{"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, +{"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, +{"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, +{"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, +{"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, +{"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, +{"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}, +{"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}, +{"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}, +{"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}, +{"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM}, +{"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}, +{"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, +{"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, +{"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, +{"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, +{"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, +{"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, +{"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, +{"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, +{"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM}, +{"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, +{"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, +{"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"}, +{"swr_flags" , NULL , 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" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM}, +{"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"}, +{"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"}, +{"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"}, +{"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"}, +{"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM }, +{"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM }, +{"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, +{"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM }, +{"min_comp" , "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" , "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" , "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" , "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 }, +{ "matrix_encoding" , "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", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, + { "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, + { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, +{ "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" }, + { "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" }, + { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" }, + { "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" }, +{ "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , 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->dither); + swri_audio_convert_free(&s-> in_convert); + swri_audio_convert_free(&s->out_convert); + swri_audio_convert_free(&s->full_convert); + swri_resample_free(&s->resample); + swri_rematrix_free(s); + } + + av_freep(ss); +} + +av_cold int swr_init(struct SwrContext *s){ + 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->dither); + 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(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_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->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ + 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); + }else + swri_resample_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; + } + + 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; + +#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) { + av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n"); + 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; + + 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); + + + 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); + } + + 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; + } +} |