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-rw-r--r--libswresample/swresample.c932
1 files changed, 932 insertions, 0 deletions
diff --git a/libswresample/swresample.c b/libswresample/swresample.c
new file mode 100644
index 0000000000..9b71b2e122
--- /dev/null
+++ b/libswresample/swresample.c
@@ -0,0 +1,932 @@
+/*
+ * 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_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; 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= 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; 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_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 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; 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;
+ }
+}