diff options
Diffstat (limited to 'libswresample/swresample.c')
-rw-r--r-- | libswresample/swresample.c | 459 |
1 files changed, 459 insertions, 0 deletions
diff --git a/libswresample/swresample.c b/libswresample/swresample.c new file mode 100644 index 0000000..c0a1893 --- /dev/null +++ b/libswresample/swresample.c @@ -0,0 +1,459 @@ +/* + * 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 ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0}, +{"och", "output channel count", OFFSET(out.ch_count ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0}, +{"isr", "input sample rate" , OFFSET( in_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0}, +{"osr", "output sample rate" , OFFSET(out_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0}, +//{"ip" , "input planar" , OFFSET( in.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0}, +//{"op" , "output planar" , OFFSET(out.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0}, +{"isf", "input sample format", OFFSET( in_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0}, +{"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0}, +{"tsf", "internal sample format", OFFSET(int_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0}, +{"icl", "input channel layout" , OFFSET( in_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"}, +{"ocl", "output channel layout", OFFSET(out_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"}, +{"clev", "center mix level" , OFFSET(clev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0}, +{"slev", "sourround mix level" , OFFSET(slev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0}, +{"flags", NULL , OFFSET(flags) , FF_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"}, +{"res", "force resampling", 0, FF_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_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; + +#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, i/*, 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 , 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; +} |