swresample.c

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/*
 * 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/mem.h"
#include "libavutil/opt.h"
#include "swresample_internal.h"
#include "audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/internal.h"
 
#include <float.h>
 
#define ALIGN 32
 
int swri_check_chlayout(struct SwrContext *s, const AVChannelLayout *chl, const char *name) {
    char l1[1024];
    int ret;
 
    if (!(ret = av_channel_layout_check(chl)) || chl->nb_channels > SWR_CH_MAX) {
        if (ret)
            av_channel_layout_describe(chl, l1, sizeof(l1));
        av_log(s, AV_LOG_WARNING, "%s channel layout \"%s\" is invalid or unsupported.\n", name, ret ? l1 : "");
        return AVERROR(EINVAL);
    }
 
    return 0;
}
 
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;
}
 
int swr_alloc_set_opts2(struct SwrContext **ps,
                        const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
                        const AVChannelLayout *in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
                        int log_offset, void *log_ctx) {
    struct SwrContext *s = *ps;
    int ret;
 
    if (!s) s = swr_alloc();
    if (!s) return AVERROR(ENOMEM);
 
    *ps = s;
 
    s->log_level_offset = log_offset;
    s->log_ctx = log_ctx;
 
    if ((ret = av_opt_set_chlayout(s, "ochl", out_ch_layout, 0)) < 0)
        goto fail;
    if ((ret = swri_check_chlayout(s, out_ch_layout, "ochl")) < 0)
        goto fail;
 
    if ((ret = av_opt_set_int(s, "osf", out_sample_fmt, 0)) < 0)
        goto fail;
 
    if ((ret = av_opt_set_int(s, "osr", out_sample_rate, 0)) < 0)
        goto fail;
 
    if ((ret = av_opt_set_chlayout(s, "ichl", in_ch_layout, 0)) < 0)
        goto fail;
    if ((ret = swri_check_chlayout(s, in_ch_layout, "ichl")) < 0)
        goto fail;
 
    if ((ret = av_opt_set_int(s, "isf", in_sample_fmt, 0)) < 0)
        goto fail;
 
    if ((ret = av_opt_set_int(s, "isr", in_sample_rate, 0)) < 0)
        goto fail;
 
    return 0;
fail:
    av_log(s, AV_LOG_ERROR, "Failed to set option\n");
    swr_free(ps);
    return ret;
}
 
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);
    if (a->ch_count == 1)
        a->planar = 1;
}
 
static void free_temp(AudioData *a){
    av_free(a->data);
    memset(a, 0, sizeof(*a));
}
 
static void clear_context(SwrContext *s){
    s->in_buffer_index= 0;
    s->in_buffer_count= 0;
    s->resample_in_constraint= 0;
    memset(s->in.ch, 0, sizeof(s->in.ch));
    memset(s->out.ch, 0, sizeof(s->out.ch));
    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);
    av_channel_layout_uninit(&s->in_ch_layout);
    av_channel_layout_uninit(&s->out_ch_layout);
    av_channel_layout_uninit(&s->used_ch_layout);
    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->delayed_samples_fixup = 0;
    s->flushed = 0;
}
 
av_cold void swr_free(SwrContext **ss){
    SwrContext *s= *ss;
    if(s){
        clear_context(s);
        av_channel_layout_uninit(&s->user_in_chlayout);
        av_channel_layout_uninit(&s->user_out_chlayout);
        av_channel_layout_uninit(&s->user_used_chlayout);
 
        if (s->resampler)
            s->resampler->free(&s->resample);
    }
 
    av_freep(ss);
}
 
av_cold void swr_close(SwrContext *s){
    clear_context(s);
}
 
av_cold int swr_init(struct SwrContext *s){
    int ret, mixing;
    char l1[1024], l2[1024];
 
    clear_context(s);
 
    if((unsigned) 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((unsigned) 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-> in_sample_rate <= 0){
        av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
        return AVERROR(EINVAL);
    }
    if(s->out_sample_rate <= 0){
        av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
        return AVERROR(EINVAL);
    }
 
    if (s->out_sample_fmt == AV_SAMPLE_FMT_DSD &&
        !(s->in_sample_fmt == AV_SAMPLE_FMT_DSD &&
          s->in_sample_rate == s->out_sample_rate &&
          !(s->flags & SWR_FLAG_RESAMPLE))) {
        av_log(s, AV_LOG_ERROR, "Conversion to DSD is not supported\n");
        return AVERROR(EINVAL);
    }
 
    s->out.ch_count  = s-> user_out_chlayout.nb_channels;
    s-> in.ch_count  = s->  user_in_chlayout.nb_channels;
 
    if (swri_check_chlayout(s, &s->user_in_chlayout , "input") ||
        swri_check_chlayout(s, &s->user_out_chlayout, "output"))
        return AVERROR(EINVAL);
 
    ret  = av_channel_layout_copy(&s->in_ch_layout, &s->user_in_chlayout);
    ret |= av_channel_layout_copy(&s->out_ch_layout, &s->user_out_chlayout);
    ret |= av_channel_layout_copy(&s->used_ch_layout, &s->user_used_chlayout);
    if (ret < 0)
        return ret;
 
    s->int_sample_fmt= s->user_int_sample_fmt;
 
    s->dither.method = s->user_dither_method;
 
    switch(s->engine){
#if CONFIG_LIBSOXR
        case SWR_ENGINE_SOXR: s->resampler = &swri_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 (!av_channel_layout_check(&s->used_ch_layout))
        av_channel_layout_default(&s->used_ch_layout, s->in.ch_count);
 
    if (s->used_ch_layout.nb_channels != s->in_ch_layout.nb_channels)
        av_channel_layout_uninit(&s->in_ch_layout);
 
    if (s->used_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
        av_channel_layout_default(&s->used_ch_layout, s->used_ch_layout.nb_channels);
    if (s->in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) {
        ret = av_channel_layout_copy(&s->in_ch_layout, &s->used_ch_layout);
        if (ret < 0)
            return ret;
    }
    if (s->out_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
        av_channel_layout_default(&s->out_ch_layout, s->out.ch_count);
 
    s->rematrix = av_channel_layout_compare(&s->out_ch_layout, &s->in_ch_layout) ||
                 s->rematrix_volume!=1.0 ||
                 s->rematrix_custom;
 
    av_channel_layout_describe(&s->out_ch_layout, l2, sizeof(l2));
    av_channel_layout_describe(&s->in_ch_layout, l1, sizeof(l1));
    if ((   s->out_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC
         || s-> in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) && s->used_ch_layout.nb_channels != s->out.ch_count && !s->rematrix_custom) {
        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 AVERROR(EINVAL);
    }
 
    /* a matrix that only copies channels leaves the samples as they are,
     * so the format in between is selected like without a matrix */
    mixing = 0;
    if (s->rematrix) {
        mixing = swri_rematrix_build(s);
        if (mixing < 0)
            return mixing;
    }
 
    if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
        // DSD to PCM conversion is done in floating point
        if(   s->in_sample_fmt == AV_SAMPLE_FMT_DSD
           && s->out_sample_fmt != AV_SAMPLE_FMT_DSD) {
            s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
        // 16bit or less to 16bit or less with the same sample rate
        } else if(   av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
           && av_get_bytes_per_sample(s->out_sample_fmt) <= 2
           && s->out_sample_rate==s->in_sample_rate) {
            s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
        // 8 -> 8, 16->8, 8->16bit
        } else if(   av_get_bytes_per_sample(s-> in_sample_fmt)
                    +av_get_bytes_per_sample(s->out_sample_fmt) <= 3 ) {
            s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
        }else if(   av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
           && !mixing
           && s->out_sample_rate==s->in_sample_rate
           && !(s->flags & SWR_FLAG_RESAMPLE)){
            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
                 && !mixing
                 && s->out_sample_rate == s->in_sample_rate
                 && !(s->flags & SWR_FLAG_RESAMPLE)
                 && s->engine != SWR_ENGINE_SOXR){
            s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
        }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
            s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
        }else{
            s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
        }
    }
    av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
 
    if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
        &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
        &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P
        &&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, s16p/s32p/s64p/fltp/dblp are 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;
        if (s->firstpts == AV_NOPTS_VALUE)
            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, s->exact_rational);
        if (!s->resample) {
            av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
            return AVERROR(ENOMEM);
        }
    }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 s16p/s32p/fltp/dblp\n");
        ret = AVERROR(EINVAL);
        goto fail;
    }
 
#define RSC 1 //FIXME finetune
    if(!s-> in.ch_count)
        s-> in.ch_count = s->in_ch_layout.nb_channels;
    if (!av_channel_layout_check(&s->used_ch_layout))
        av_channel_layout_default(&s->used_ch_layout, s->in.ch_count);
    if(!s->out.ch_count)
        s->out.ch_count = s->out_ch_layout.nb_channels;
 
    if(!s-> in.ch_count){
        av_assert0(s->in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC);
        av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
        ret = AVERROR(EINVAL);
        goto fail;
    }
 
    if (s->in_ch_layout.order != AV_CHANNEL_ORDER_UNSPEC && s->used_ch_layout.nb_channels != s->in_ch_layout.nb_channels) {
        av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_layout.nb_channels);
        ret = AVERROR(EINVAL);
        goto fail;
    }
 
av_assert0(s->used_ch_layout.nb_channels);
av_assert0(s->out.ch_count);
    s->resample_first= RSC*s->out.ch_count/s->used_ch_layout.nb_channels - 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 ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
        goto fail;
 
    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);
        // fall through to the generic path for conversions that have no
        // direct implementation (e.g. DSD input to non-float output)
        if (s->full_convert)
            return 0;
    }
 
    s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
                                             s-> in_sample_fmt, s->used_ch_layout.nb_channels, 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) {
        av_log(s, AV_LOG_ERROR, "Cannot convert %s sample format to %s sample format\n",
               av_get_sample_fmt_name(!s->in_convert ? s->in_sample_fmt : s->int_sample_fmt),
               av_get_sample_fmt_name(!s->in_convert ? s->int_sample_fmt : s->out_sample_fmt));
        ret = AVERROR(EINVAL);
        goto fail;
    }
 
    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_layout.nb_channels;
        if(s->resample)
            s->in_buffer.ch_count= s->used_ch_layout.nb_channels;
    }
    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);
    }
 
    av_assert0(!s->preout.count);
    s->dither.noise = s->preout;
    s->dither.temp  = s->preout;
    if (s->dither.method > SWR_DITHER_NS) {
        s->dither.noise.bps = 4;
        s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP;
        s->dither.noise_scale = 1;
    }
 
    if(s->rematrix || s->dither.method) {
        ret = swri_rematrix_init(s);
        if (ret < 0)
            goto fail;
    }
 
    return 0;
fail:
    swr_close(s);
    return ret;
 
}
 
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_calloc(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->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
    }
    if(a->count && !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 *const 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;
    int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 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;
 
    border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
                 &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
    if (border == INT_MAX) {
        return 0;
    } else if (border < 0) {
        return border;
    } else if (border) {
        buf_set(&in, &in, border);
        in_count -= border;
        s->resample_in_constraint = 0;
    }
 
    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 > padless) && !s->in_buffer_count){
            s->in_buffer_index=0;
            ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &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;
            if (padless) {
                padless = 0;
                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_layout.nb_channels);
        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)
            if ((out_count = resample(s, midbuf, out_count, postin, in_count)) < 0)
                return out_count;
        if(midbuf != preout && out_count)
            swri_rematrix(s, preout, midbuf, out_count, preout==out);
    }else{
        if(postin != midbuf && in_count)
            swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
        if(midbuf != preout)
            if ((out_count = resample(s, preout, out_count, midbuf, in_count)) < 0)
                return out_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++)
                    if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
                        return ret;
            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, &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_is_initialized(struct SwrContext *s) {
    return !!s->in_buffer.ch_count;
}
 
int attribute_align_arg swr_convert(struct SwrContext *s,
                                          uint8_t * const *out_arg, int out_count,
                                    const uint8_t * const *in_arg,  int in_count)
{
    AudioData * in= &s->in;
    AudioData *out= &s->out;
 
    if (!swr_is_initialized(s)) {
        av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
        return AVERROR(EINVAL);
    }
#if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
    int max_output = swr_get_out_samples(s, in_count);
#endif
 
    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;
            if (!s->drop_output && !out_arg)
                return 0;
            continue;
        }
 
        av_assert0(s->drop_output);
        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;
 
        av_assert2(max_output < 0 || ret <= max_output);
 
        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;
        av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
        return ret2;
    }
}
 
int swr_drop_output(struct SwrContext *s, int count){
    const uint8_t *tmp_arg[SWR_CH_MAX];
    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, tmp_arg, 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;
 
    {
        int fill = s->silence.fmt == AV_SAMPLE_FMT_DSD ? 0x69 :
                   s->silence.bps == 1                 ? 0x80 : 0;
        if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
            memset(s->silence.ch[i], fill, count*s->silence.bps);
        } else
            memset(s->silence.ch[0], fill, 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_get_out_samples(struct SwrContext *s, int in_samples)
{
    int64_t out_samples;
 
    if (in_samples < 0)
        return AVERROR(EINVAL);
 
    if (s->resampler && s->resample) {
        if (!s->resampler->get_out_samples)
            return AVERROR(ENOSYS);
        out_samples = s->resampler->get_out_samples(s, in_samples);
    } else {
        out_samples = s->in_buffer_count + in_samples;
        av_assert0(s->out_sample_rate == s->in_sample_rate);
    }
 
    if (out_samples > INT_MAX)
        return AVERROR(EINVAL);
 
    return out_samples;
}
 
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;
    }
}