1 files changed, 460 insertions, 0 deletions

diff --git a/libswresample/swresample.c b/libswresample/swresample.c
new file mode 100644
index 0000000..d0fb96e
--- /dev/null
+++ b/libswresample/swresample.c
@@ -0,0 +1,460 @@
+/*
+ * Copyright (C) 2011 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"
+
+#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
+
+
+//TODO split options array out?
+#define OFFSET(x) offsetof(SwrContext,x)
+static const AVOption options[]={
+{"ich",  "input channel count", OFFSET( in.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
+{"och", "output channel count", OFFSET(out.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
+{"isr",  "input sample rate"  , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
+{"osr", "output sample rate"  , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
+//{"ip" ,  "input planar"       , OFFSET( in.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
+//{"op" , "output planar"       , OFFSET(out.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
+{"isf",  "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
+{"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
+{"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
+{"icl",  "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
+{"ocl",  "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
+{"clev", "center mix level"     , OFFSET(clev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
+{"slev", "sourround mix level"  , OFFSET(slev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
+{"rmvol", "rematrix volume"     , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
+{"flags", NULL                  , OFFSET(flags)        , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
+{"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
+
+{0}
+};
+
+static const char* context_to_name(void* ptr) {
+    return "SWR";
+}
+
+static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
+
+static int resample(SwrContext *s, AudioData *out_param, int out_count,
+                             const AudioData * in_param, int in_count);
+
+SwrContext *swr_alloc(void){
+    SwrContext *s= av_mallocz(sizeof(SwrContext));
+    if(s){
+        s->av_class= &av_class;
+        av_opt_set_defaults2(s, 0, 0);
+    }
+    return s;
+}
+
+SwrContext *swr_alloc2(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_set_int(s, "ocl", out_ch_layout);
+    av_set_int(s, "osf", out_sample_fmt);
+    av_set_int(s, "osr", out_sample_rate);
+    av_set_int(s, "icl", in_ch_layout);
+    av_set_int(s, "isf", in_sample_fmt);
+    av_set_int(s, "isr", in_sample_rate);
+
+    s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
+    s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
+    s->int_sample_fmt = AV_SAMPLE_FMT_S16;
+
+    return s;
+}
+
+
+static void free_temp(AudioData *a){
+    av_free(a->data);
+    memset(a, 0, sizeof(*a));
+}
+
+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);
+        swr_audio_convert_free(&s-> in_convert);
+        swr_audio_convert_free(&s->out_convert);
+        swr_audio_convert_free(&s->full_convert);
+        swr_resample_free(&s->resample);
+    }
+
+    av_freep(ss);
+}
+
+int swr_init(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);
+    swr_audio_convert_free(&s-> in_convert);
+    swr_audio_convert_free(&s->out_convert);
+    swr_audio_convert_free(&s->full_convert);
+
+    s-> in.planar= s-> in_sample_fmt >= 0x100;
+    s->out.planar= s->out_sample_fmt >= 0x100;
+    s-> in_sample_fmt &= 0xFF;
+    s->out_sample_fmt &= 0xFF;
+
+    if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
+        av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
+        return AVERROR(EINVAL);
+    }
+    if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
+        av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
+        return AVERROR(EINVAL);
+    }
+
+    if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16
+        &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
+        av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
+        return AVERROR(EINVAL);
+    }
+
+    //FIXME should we allow/support using FLT on material that doesnt need it ?
+    if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
+        s->int_sample_fmt= AV_SAMPLE_FMT_S16;
+    }else
+        s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
+
+
+    if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
+        s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
+    }else
+        swr_resample_free(&s->resample);
+    if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
+        av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
+        return -1;
+    }
+
+    if(s-> in.ch_count && s-> in_ch_layout && s->in.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 there actually is, ignoring layout\n");
+        s-> in_ch_layout= 0;
+    }
+
+    if(!s-> in_ch_layout)
+        s-> in_ch_layout= av_get_default_channel_layout(s->in.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;
+
+#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->out.ch_count)
+        s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
+
+av_assert0(s-> in.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.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
+    s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
+    s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
+
+    if(!s->resample && !s->rematrix){
+        s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
+                                                  s-> in_sample_fmt, s-> in.ch_count, 0);
+        return 0;
+    }
+
+    s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
+                                            s-> in_sample_fmt, s-> in.ch_count, 0);
+    s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
+                                            s->int_sample_fmt, s->out.ch_count, 0);
+
+
+    s->postin= s->in;
+    s->preout= s->out;
+    s->midbuf= s->in;
+    s->in_buffer= s->in;
+    if(!s->resample_first){
+        s->midbuf.ch_count= s->out.ch_count;
+        s->in_buffer.ch_count = s->out.ch_count;
+    }
+
+    s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
+    s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;
+
+
+    if(s->rematrix && swr_rematrix_init(s)<0)
+        return -1;
+
+    return 0;
+}
+
+static int realloc_audio(AudioData *a, int count){
+    int i, countb;
+    AudioData old;
+
+    if(a->count >= count)
+        return 0;
+
+    count*=2;
+
+    countb= FFALIGN(count*a->bps, 32);
+    old= *a;
+
+    av_assert0(a->planar);
+    av_assert0(a->bps);
+    av_assert0(a->ch_count);
+
+    a->data= av_malloc(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);
+    }
+    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(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;
+    }
+}
+
+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 *postin, *midbuf, *preout;
+    int ret/*, in_max*/;
+    AudioData * in= &s->in;
+    AudioData *out= &s->out;
+    AudioData preout_tmp, midbuf_tmp;
+
+    if(!s->resample){
+        if(in_count > out_count)
+            return -1;
+        out_count = in_count;
+    }
+
+    fill_audiodata(in ,  (void*)in_arg);
+    fill_audiodata(out, out_arg);
+
+    if(s->full_convert){
+        av_assert0(!s->resample);
+        swr_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-> in.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)
+        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 teh 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){
+        swr_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)
+            swr_rematrix(s, preout, midbuf, out_count, preout==out);
+    }else{
+        if(postin != midbuf)
+            swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
+        if(midbuf != preout)
+            out_count= resample(s, preout, out_count, midbuf, in_count);
+    }
+
+    if(preout != out){
+//FIXME packed doesnt need more than 1 chan here!
+        swr_audio_convert(s->out_convert, out, preout, out_count);
+    }
+    return out_count;
+}
+
+/**
+ *
+ * out may be equal in.
+ */
+static void buf_set(AudioData *out, AudioData *in, int count){
+    if(in->planar){
+        int ch;
+        for(ch=0; ch<out->ch_count; ch++)
+            out->ch[ch]= in->ch[ch] + count*out->bps;
+    }else
+        out->ch[0]= in->ch[0] + 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;
+
+    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= swr_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= swr_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;
+}