September 2011 - ffmpeg-cvslog

avfiltergraph: change the syntax of avfilter_graph_parse()
by git＠videolan.org 18 May '12

18 May '12

ffmpeg | branch: master | Stefano Sabatini <stefano.sabatini-lala(a)poste.it> | Sat Jun 11 15:16:02 2011 +0200| [6119b23a3662d1e106cdf69ef3171b2e7e1d495c] | committer: Stefano Sabatini avfiltergraph: change the syntax of avfilter_graph_parse() Make it returns the list of open inputs and outputs, so it can be reused by applications. Breaks API/ABI. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6119b23a3662d1e10… --- doc/APIchanges | 3 +++ ffmpeg.c | 2 +- ffplay.c | 2 +- libavfilter/avfilter.h | 4 ++-- libavfilter/avfiltergraph.h | 8 +++++--- libavfilter/graphparser.c | 20 ++++++++++---------- 6 files changed, 22 insertions(+), 17 deletions(-) diff --git a/doc/APIchanges b/doc/APIchanges index 592c268..40efee8 100644 --- a/doc/APIchanges +++ b/doc/APIchanges @@ -13,6 +13,9 @@ libavutil: 2011-04-18 API changes, most recent first: +2011-06-12 - xxxxxxx - lavfi 2.16.0 - avfilter_graph_parse() + Change avfilter_graph_parse() signature. + 2011-06-xx - xxxxxxx - lavu 51.6.0 - opt.h Add av_opt_flag_is_set(). diff --git a/ffmpeg.c b/ffmpeg.c index 35395a9..f69c054 100644 --- a/ffmpeg.c +++ b/ffmpeg.c @@ -418,7 +418,7 @@ static int configure_video_filters(AVInputStream *ist, AVOutputStream *ost) inputs->pad_idx = 0; inputs->next = NULL; - if ((ret = avfilter_graph_parse(ost->graph, ost->avfilter, inputs, outputs, NULL)) < 0) + if ((ret = avfilter_graph_parse(ost->graph, ost->avfilter, &inputs, &outputs, NULL)) < 0) return ret; av_freep(&ost->avfilter); } else { diff --git a/ffplay.c b/ffplay.c index ebe9e05..c5f5f75 100644 --- a/ffplay.c +++ b/ffplay.c @@ -1708,7 +1708,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c inputs->pad_idx = 0; inputs->next = NULL; - if ((ret = avfilter_graph_parse(graph, vfilters, inputs, outputs, NULL)) < 0) + if ((ret = avfilter_graph_parse(graph, vfilters, &inputs, &outputs, NULL)) < 0) goto the_end; av_freep(&vfilters); } else { diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h index 64d76cc..fbd1dc4 100644 --- a/libavfilter/avfilter.h +++ b/libavfilter/avfilter.h @@ -26,8 +26,8 @@ #include "libavutil/samplefmt.h" #define LIBAVFILTER_VERSION_MAJOR 2 -#define LIBAVFILTER_VERSION_MINOR 15 -#define LIBAVFILTER_VERSION_MICRO 1 +#define LIBAVFILTER_VERSION_MINOR 16 +#define LIBAVFILTER_VERSION_MICRO 0 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \ diff --git a/libavfilter/avfiltergraph.h b/libavfilter/avfiltergraph.h index 23a7c51..538fd2c 100644 --- a/libavfilter/avfiltergraph.h +++ b/libavfilter/avfiltergraph.h @@ -112,12 +112,14 @@ typedef struct AVFilterInOut { * * @param graph the filter graph where to link the parsed graph context * @param filters string to be parsed - * @param inputs linked list to the inputs of the graph - * @param outputs linked list to the outputs of the graph + * @param inputs linked list to the inputs of the graph, may be NULL. + * It is updated to contain the list of open inputs after the parsing. + * @param outputs linked list to the outputs of the graph, may be NULL. + * It is updated to contain the list of open outputs after the parsing. * @return zero on success, a negative AVERROR code on error */ int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, - AVFilterInOut *inputs, AVFilterInOut *outputs, + AVFilterInOut **inputs, AVFilterInOut **outputs, void *log_ctx); #endif /* AVFILTER_AVFILTERGRAPH_H */ diff --git a/libavfilter/graphparser.c b/libavfilter/graphparser.c index aa1f228..ea0c5dd 100644 --- a/libavfilter/graphparser.c +++ b/libavfilter/graphparser.c @@ -328,8 +328,8 @@ static int parse_outputs(const char **buf, AVFilterInOut **curr_inputs, } int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, - AVFilterInOut *open_inputs, - AVFilterInOut *open_outputs, void *log_ctx) + AVFilterInOut **open_inputs, AVFilterInOut **open_outputs, + void *log_ctx) { int index = 0, ret; char chr = 0; @@ -341,7 +341,7 @@ int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, const char *filterchain = filters; filters += strspn(filters, WHITESPACES); - if ((ret = parse_inputs(&filters, &curr_inputs, &open_outputs, log_ctx)) < 0) + if ((ret = parse_inputs(&filters, &curr_inputs, open_outputs, log_ctx)) < 0) goto fail; if ((ret = parse_filter(&filter, &filters, graph, index, log_ctx)) < 0) @@ -350,14 +350,14 @@ int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, if (filter->input_count == 1 && !curr_inputs && !index) { /* First input can be omitted if it is "[in]" */ const char *tmp = "[in]"; - if ((ret = parse_inputs(&tmp, &curr_inputs, &open_outputs, log_ctx)) < 0) + if ((ret = parse_inputs(&tmp, &curr_inputs, open_outputs, log_ctx)) < 0) goto fail; } - if ((ret = link_filter_inouts(filter, &curr_inputs, &open_inputs, log_ctx)) < 0) + if ((ret = link_filter_inouts(filter, &curr_inputs, open_inputs, log_ctx)) < 0) goto fail; - if ((ret = parse_outputs(&filters, &curr_inputs, &open_inputs, &open_outputs, + if ((ret = parse_outputs(&filters, &curr_inputs, open_inputs, open_outputs, log_ctx)) < 0) goto fail; @@ -382,10 +382,10 @@ int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, goto fail; } - if (open_inputs && !strcmp(open_inputs->name, "out") && curr_inputs) { + if (*open_inputs && !strcmp((*open_inputs)->name, "out") && curr_inputs) { /* Last output can be omitted if it is "[out]" */ const char *tmp = "[out]"; - if ((ret = parse_outputs(&tmp, &curr_inputs, &open_inputs, &open_outputs, + if ((ret = parse_outputs(&tmp, &curr_inputs, open_inputs, open_outputs, log_ctx)) < 0) goto fail; } @@ -396,8 +396,8 @@ int avfilter_graph_parse(AVFilterGraph *graph, const char *filters, for (; graph->filter_count > 0; graph->filter_count--) avfilter_free(graph->filters[graph->filter_count - 1]); av_freep(&graph->filters); - free_inout(open_inputs); - free_inout(open_outputs); + free_inout(*open_inputs); + free_inout(*open_outputs); free_inout(curr_inputs); return ret; }

2 1

Add libswresample.
by git＠videolan.org 07 Oct '11

07 Oct '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Mon Sep 19 06:13:30 2011 +0200| [b5875b91113a0f3de6ad61e9ff8b74b81de94760] | committer: Michael Niedermayer Add libswresample. Similar to libswscale this does resampling and format convertion, just for audio instead of video. changing sampling rate, sample formats, channel layouts and sample packing all in one with a very simple public interface. Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b5875b91113a0f3de… --- Makefile | 1 + common.mak | 2 +- configure | 7 +- ffmpeg.c | 83 +++----- libswresample/Makefile | 12 + libswresample/audioconvert.c | 113 +++++++++ libswresample/audioconvert.h | 65 ++++++ libswresample/rematrix.c | 271 ++++++++++++++++++++++ libswresample/rematrix_template.c | 38 +++ libswresample/resample2.c | 352 ++++++++++++++++++++++++++++ libswresample/swresample.c | 432 +++++++++++++++++++++++++++++++++++ libswresample/swresample.h | 79 +++++++ libswresample/swresample_internal.h | 77 ++++++ libswresample/swresample_test.c | 149 ++++++++++++ tests/ref/acodec/g726 | 6 +- tests/ref/acodec/pcm | 6 +- tests/ref/lavf/dv_fmt | 4 +- tests/ref/lavf/mxf_d10 | 4 +- 18 files changed, 1634 insertions(+), 67 deletions(-) diff --git a/Makefile b/Makefile index 27b841f..167dd32 100644 --- a/Makefile +++ b/Makefile @@ -33,6 +33,7 @@ FFLIBS-$(CONFIG_AVFILTER) += avfilter FFLIBS-$(CONFIG_AVFORMAT) += avformat FFLIBS-$(CONFIG_AVCODEC) += avcodec FFLIBS-$(CONFIG_POSTPROC) += postproc +FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample FFLIBS-$(CONFIG_SWSCALE) += swscale FFLIBS := avutil diff --git a/common.mak b/common.mak index bad86fd..8dd2439 100644 --- a/common.mak +++ b/common.mak @@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR)))) $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL)) endif -ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale +ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample # NASM requires -I path terminated with / IFLAGS := -I. -I$(SRC_PATH)/ diff --git a/configure b/configure index 4cdb491..dfc8ee3 100755 --- a/configure +++ b/configure @@ -89,6 +89,7 @@ Configuration options: --disable-avdevice disable libavdevice build --disable-avcodec disable libavcodec build --disable-avformat disable libavformat build + --disable-swresample disable libswresample build --disable-swscale disable libswscale build --disable-postproc disable libpostproc build --disable-avfilter disable video filter support [no] @@ -1037,6 +1038,7 @@ CONFIG_LIST=" small sram static + swresample swscale swscale_alpha thumb @@ -1603,7 +1605,7 @@ avformat_deps="avcodec" postproc_deps="gpl" # programs -ffmpeg_deps="avcodec avformat swscale" +ffmpeg_deps="avcodec avformat swscale swresample" ffmpeg_select="buffer_filter buffersink_filter" avconv_deps="avcodec avformat swscale" avconv_select="buffer_filter" @@ -1766,6 +1768,7 @@ enable postproc enable protocols enable static enable stripping +enable swresample enable swscale enable swscale_alpha @@ -3143,7 +3146,7 @@ enabled extra_warnings && check_cflags -Winline # add some linker flags check_ldflags -Wl,--warn-common -check_ldflags -Wl,-rpath-link=libpostproc:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil +check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic echo "X{};" > $TMPV diff --git a/ffmpeg.c b/ffmpeg.c index b887abd..6da97d2 100644 --- a/ffmpeg.c +++ b/ffmpeg.c @@ -45,6 +45,7 @@ #include "libavutil/avstring.h" #include "libavutil/libm.h" #include "libavformat/os_support.h" +#include "libswresample/swresample.h" #include "libavformat/ffm.h" // not public API @@ -229,15 +230,14 @@ typedef struct OutputStream { /* audio only */ int audio_resample; - ReSampleContext *resample; /* for audio resampling */ int resample_sample_fmt; int resample_channels; int resample_sample_rate; - int reformat_pair; - AVAudioConvert *reformat_ctx; AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */ FILE *logfile; + struct SwrContext *swr; + #if CONFIG_AVFILTER AVFilterContext *output_video_filter; AVFilterContext *input_video_filter; @@ -843,14 +843,15 @@ need_realloc: exit_program(1); } - if (enc->channels != dec->channels) + if (enc->channels != dec->channels + || enc->sample_fmt != dec->sample_fmt) ost->audio_resample = 1; resample_changed = ost->resample_sample_fmt != dec->sample_fmt || ost->resample_channels != dec->channels || ost->resample_sample_rate != dec->sample_rate; - if ((ost->audio_resample && !ost->resample) || resample_changed) { + if ((ost->audio_resample && !ost->swr) || resample_changed) { if (resample_changed) { av_log(NULL, AV_LOG_INFO, "Input stream #%d.%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n", ist->file_index, ist->st->index, @@ -859,24 +860,29 @@ need_realloc: ost->resample_sample_fmt = dec->sample_fmt; ost->resample_channels = dec->channels; ost->resample_sample_rate = dec->sample_rate; - if (ost->resample) - audio_resample_close(ost->resample); + swr_free(&ost->swr); } /* if audio_sync_method is >1 the resampler is needed for audio drift compensation */ if (audio_sync_method <= 1 && ost->resample_sample_fmt == enc->sample_fmt && ost->resample_channels == enc->channels && ost->resample_sample_rate == enc->sample_rate) { - ost->resample = NULL; + //ost->swr = NULL; ost->audio_resample = 0; } else { - if (dec->sample_fmt != AV_SAMPLE_FMT_S16) - fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n"); - ost->resample = av_audio_resample_init(enc->channels, dec->channels, - enc->sample_rate, dec->sample_rate, - enc->sample_fmt, dec->sample_fmt, - 16, 10, 0, 0.8); - if (!ost->resample) { + ost->swr = swr_alloc2(ost->swr, + enc->channel_layout, enc->sample_fmt, enc->sample_rate, + dec->channel_layout, dec->sample_fmt, dec->sample_rate, + 0, NULL); + av_set_int(ost->swr, "ich", dec->channels); + av_set_int(ost->swr, "och", enc->channels); + if(audio_sync_method>1) av_set_int(ost->swr, "flags", SWR_FLAG_RESAMPLE); + if(ost->swr && swr_init(ost->swr) < 0){ + fprintf(stderr, "swr_init() failed\n"); + swr_free(&ost->swr); + } + + if (!ost->swr) { fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n", dec->channels, dec->sample_rate, enc->channels, enc->sample_rate); @@ -885,21 +891,7 @@ need_realloc: } } -#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b)) - if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt && - MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) { - if (ost->reformat_ctx) - av_audio_convert_free(ost->reformat_ctx); - ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1, - dec->sample_fmt, 1, NULL, 0); - if (!ost->reformat_ctx) { - fprintf(stderr, "Cannot convert %s sample format to %s sample format\n", - av_get_sample_fmt_name(dec->sample_fmt), - av_get_sample_fmt_name(enc->sample_fmt)); - exit_program(1); - } - ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt); - } + av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt); if(audio_sync_method){ double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts @@ -941,7 +933,7 @@ need_realloc: if(verbose > 2) fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate); // fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2)); - av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate); + swr_compensate(ost->swr, comp, enc->sample_rate); } } }else @@ -950,30 +942,15 @@ need_realloc: if (ost->audio_resample) { buftmp = audio_buf; - size_out = audio_resample(ost->resample, - (short *)buftmp, (short *)buf, - size / (dec->channels * isize)); + size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, audio_buf_size / (enc->channels * osize), + (const uint8_t*[]){buf }, size / (dec->channels * isize)); size_out = size_out * enc->channels * osize; } else { buftmp = buf; size_out = size; } - if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) { - const void *ibuf[6]= {buftmp}; - void *obuf[6]= {audio_buf}; - int istride[6]= {isize}; - int ostride[6]= {osize}; - int len= size_out/istride[0]; - if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) { - printf("av_audio_convert() failed\n"); - if (exit_on_error) - exit_program(1); - return; - } - buftmp = audio_buf; - size_out = len*osize; - } + av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt); /* now encode as many frames as possible */ if (enc->frame_size > 1) { @@ -2133,7 +2110,6 @@ static int transcode_init(OutputFile *output_files, int nb_output_files, if (!ost->fifo) { return AVERROR(ENOMEM); } - ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE); if (!codec->sample_rate) { codec->sample_rate = icodec->sample_rate; } @@ -2149,6 +2125,8 @@ static int transcode_init(OutputFile *output_files, int nb_output_files, if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels) codec->channel_layout = 0; ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1; + ost->audio_resample |= codec->sample_fmt != icodec->sample_fmt + || codec->channel_layout != icodec->channel_layout; icodec->request_channels = codec->channels; ist->decoding_needed = 1; ost->encoding_needed = 1; @@ -2679,10 +2657,7 @@ static int transcode(OutputFile *output_files, int nb_output_files, av_free(ost->forced_kf_pts); if (ost->video_resample) sws_freeContext(ost->img_resample_ctx); - if (ost->resample) - audio_resample_close(ost->resample); - if (ost->reformat_ctx) - av_audio_convert_free(ost->reformat_ctx); + swr_free(&ost->swr); av_dict_free(&ost->opts); } } diff --git a/libswresample/Makefile b/libswresample/Makefile new file mode 100644 index 0000000..33b8ea0 --- /dev/null +++ b/libswresample/Makefile @@ -0,0 +1,12 @@ +include $(SUBDIR)../config.mak + +NAME = swresample +FFLIBS = avutil + +HEADERS = swresample.h + +OBJS = swresample.o audioconvert.o resample2.o rematrix.o + +TESTPROGS = swresample_test + +include $(SUBDIR)../subdir.mak diff --git a/libswresample/audioconvert.c b/libswresample/audioconvert.c new file mode 100644 index 0000000..a1fa3eb --- /dev/null +++ b/libswresample/audioconvert.c @@ -0,0 +1,113 @@ +/* + * audio conversion + * Copyright (c) 2006 Michael Niedermayer <michaelni(a)gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * audio conversion + * @author Michael Niedermayer <michaelni(a)gmx.at> + */ + +#include "libavutil/avstring.h" +#include "libavutil/avassert.h" +#include "libavutil/libm.h" +#include "libavutil/samplefmt.h" +#include "audioconvert.h" + + +struct AVAudioConvert { + int channels; + int fmt_pair; +}; + +AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt, + enum AVSampleFormat in_fmt, + int channels, int flags) +{ + AVAudioConvert *ctx; + ctx = av_malloc(sizeof(AVAudioConvert)); + if (!ctx) + return NULL; + ctx->channels = channels; + ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt; + return ctx; +} + +void swr_audio_convert_free(AVAudioConvert **ctx) +{ + av_freep(ctx); +} + +int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData*in, int len) +{ + int ch; + + av_assert0(ctx->channels == out->ch_count); + + //FIXME optimize common cases + + for(ch=0; ch<ctx->channels; ch++){ + const int is= (in ->planar ? 1 : in->ch_count) * in->bps; + const int os= (out->planar ? 1 :out->ch_count) *out->bps; + const uint8_t *pi= in ->ch[ch]; + uint8_t *po= out->ch[ch]; + uint8_t *end= po + os*len; + if(!po) + continue; + +#define CONV(ofmt, otype, ifmt, expr)\ +if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\ + do{\ + *(otype*)po = expr; pi += is; po += os;\ + }while(po < end);\ +} + +//FIXME put things below under ifdefs so we do not waste space for cases no codec will need +//FIXME rounding ? + + CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi) + else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) + else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) + else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) + else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) + else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) + else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi) + else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16) + else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) + else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) + else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) + else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16) + else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi) + else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) + else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) + else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) + else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) + else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31)))) + else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi) + else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi) + else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80)) + else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15)))) + else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31)))) + else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi) + else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi) + else return -1; + } + return 0; +} diff --git a/libswresample/audioconvert.h b/libswresample/audioconvert.h new file mode 100644 index 0000000..e5fd4df --- /dev/null +++ b/libswresample/audioconvert.h @@ -0,0 +1,65 @@ +/* + * audio conversion + * Copyright (c) 2006 Michael Niedermayer <michaelni(a)gmx.at> + * Copyright (c) 2008 Peter Ross + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef SWR_AUDIOCONVERT_H +#define SWR_AUDIOCONVERT_H + +/** + * @file + * Audio format conversion routines + */ + + +#include "swresample_internal.h" +#include "libavutil/cpu.h" +#include "libavutil/audioconvert.h" + +struct AVAudioConvert; +typedef struct AVAudioConvert AVAudioConvert; + +/** + * Create an audio sample format converter context + * @param out_fmt Output sample format + * @param in_fmt Input sample format + * @param channels Number of channels + * @param flags See AV_CPU_FLAG_xx + * @return NULL on error + */ +AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt, + enum AVSampleFormat in_fmt, + int channels, int flags); + +/** + * Free audio sample format converter context. + * and set the pointer to NULL + */ +void swr_audio_convert_free(AVAudioConvert **ctx); + +/** + * Convert between audio sample formats + * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel. + * @param[in] in array of input buffers for each channel + * @param len length of audio frame size (measured in samples) + */ +int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData *in, int len); + +#endif /* AVCODEC_AUDIOCONVERT_H */ diff --git a/libswresample/rematrix.c b/libswresample/rematrix.c new file mode 100644 index 0000000..b8c5d45 --- /dev/null +++ b/libswresample/rematrix.c @@ -0,0 +1,271 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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 "swresample_internal.h" +#include "libavutil/audioconvert.h" +#include "libavutil/avassert.h" + +#define SAMPLE float +#define RENAME(x) x ## _float +#include "rematrix_template.c" +#undef SAMPLE +#undef RENAME + +#define SAMPLE int16_t +#define RENAME(x) x ## _s16 +#include "rematrix_template.c" + + +#define FRONT_LEFT 0 +#define FRONT_RIGHT 1 +#define FRONT_CENTER 2 +#define LOW_FREQUENCY 3 +#define BACK_LEFT 4 +#define BACK_RIGHT 5 +#define FRONT_LEFT_OF_CENTER 6 +#define FRONT_RIGHT_OF_CENTER 7 +#define BACK_CENTER 8 +#define SIDE_LEFT 9 +#define SIDE_RIGHT 10 +#define TOP_CENTER 11 +#define TOP_FRONT_LEFT 12 +#define TOP_FRONT_CENTER 13 +#define TOP_FRONT_RIGHT 14 +#define TOP_BACK_LEFT 15 +#define TOP_BACK_CENTER 16 +#define TOP_BACK_RIGHT 17 + +static int even(int64_t layout){ + if(!layout) return 1; + if(layout&(layout-1)) return 1; + return 0; +} + +static int sane_layout(int64_t layout){ + if(!(layout & AV_CH_LAYOUT_SURROUND)) // at least 1 front speaker + return 0; + if(!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT))) // no asymetric front + return 0; + if(!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT))) // no asymetric side + return 0; + if(!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT))) + return 0; + if(!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER))) + return 0; + if(av_get_channel_layout_nb_channels(layout) >= SWR_CH_MAX) + return 0; + + return 1; +} + +int swr_rematrix_init(SwrContext *s){ + int i, j, in_i, out_i; + float matrix[64][64]={0}; + int64_t unaccounted= s->in_ch_layout & ~s->out_ch_layout; + float maxcoef=0; + + for(i=0; i<64; i++){ + if(s->in_ch_layout & s->out_ch_layout & (1LL<<i)) + matrix[i][i]= 1.0; + } + + if(!sane_layout(s->in_ch_layout)){ + av_log(s, AV_LOG_ERROR, "Input channel layout isnt supported\n"); + return AVERROR(EINVAL); + } + if(!sane_layout(s->out_ch_layout)){ + av_log(s, AV_LOG_ERROR, "Output channel layout isnt supported\n"); + return AVERROR(EINVAL); + } + +//FIXME implement dolby surround +//FIXME implement full ac3 + + + if(unaccounted & AV_CH_FRONT_CENTER){ + if((s->out_ch_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO){ + matrix[ FRONT_LEFT][FRONT_CENTER]+= sqrt(0.5); + matrix[FRONT_RIGHT][FRONT_CENTER]+= sqrt(0.5); + }else + av_assert0(0); + } + if(unaccounted & AV_CH_LAYOUT_STEREO){ + if(s->out_ch_layout & AV_CH_FRONT_CENTER){ + matrix[FRONT_CENTER][ FRONT_LEFT]+= sqrt(0.5); + matrix[FRONT_CENTER][FRONT_RIGHT]+= sqrt(0.5); + if(s->in_ch_layout & AV_CH_FRONT_CENTER) + matrix[FRONT_CENTER][ FRONT_CENTER] = s->clev*sqrt(2); + }else + av_assert0(0); + } + + if(unaccounted & AV_CH_BACK_CENTER){ + if(s->out_ch_layout & AV_CH_BACK_LEFT){ + matrix[ BACK_LEFT][BACK_CENTER]+= sqrt(0.5); + matrix[BACK_RIGHT][BACK_CENTER]+= sqrt(0.5); + }else if(s->out_ch_layout & AV_CH_SIDE_LEFT){ + matrix[ SIDE_LEFT][BACK_CENTER]+= sqrt(0.5); + matrix[SIDE_RIGHT][BACK_CENTER]+= sqrt(0.5); + }else if(s->out_ch_layout & AV_CH_FRONT_LEFT){ + matrix[ FRONT_LEFT][BACK_CENTER]+= s->slev*sqrt(0.5); + matrix[FRONT_RIGHT][BACK_CENTER]+= s->slev*sqrt(0.5); + }else if(s->out_ch_layout & AV_CH_FRONT_CENTER){ + matrix[ FRONT_CENTER][BACK_CENTER]+= s->slev*sqrt(0.5); + }else + av_assert0(0); + } + if(unaccounted & AV_CH_BACK_LEFT){ + if(s->out_ch_layout & AV_CH_BACK_CENTER){ + matrix[BACK_CENTER][ BACK_LEFT]+= sqrt(0.5); + matrix[BACK_CENTER][BACK_RIGHT]+= sqrt(0.5); + }else if(s->out_ch_layout & AV_CH_SIDE_LEFT){ + if(s->in_ch_layout & AV_CH_SIDE_LEFT){ + matrix[ SIDE_LEFT][ BACK_LEFT]+= sqrt(0.5); + matrix[SIDE_RIGHT][BACK_RIGHT]+= sqrt(0.5); + }else{ + matrix[ SIDE_LEFT][ BACK_LEFT]+= 1.0; + matrix[SIDE_RIGHT][BACK_RIGHT]+= 1.0; + } + }else if(s->out_ch_layout & AV_CH_FRONT_LEFT){ + matrix[ FRONT_LEFT][ BACK_LEFT]+= s->slev; + matrix[FRONT_RIGHT][BACK_RIGHT]+= s->slev; + }else if(s->out_ch_layout & AV_CH_FRONT_CENTER){ + matrix[ FRONT_CENTER][BACK_LEFT ]+= s->slev*sqrt(0.5); + matrix[ FRONT_CENTER][BACK_RIGHT]+= s->slev*sqrt(0.5); + }else + av_assert0(0); + } + + if(unaccounted & AV_CH_SIDE_LEFT){ + if(s->out_ch_layout & AV_CH_BACK_LEFT){ + matrix[ BACK_LEFT][ SIDE_LEFT]+= 1.0; + matrix[BACK_RIGHT][SIDE_RIGHT]+= 1.0; + }else if(s->out_ch_layout & AV_CH_BACK_CENTER){ + matrix[BACK_CENTER][ SIDE_LEFT]+= sqrt(0.5); + matrix[BACK_CENTER][SIDE_RIGHT]+= sqrt(0.5); + }else if(s->out_ch_layout & AV_CH_FRONT_LEFT){ + matrix[ FRONT_LEFT][ SIDE_LEFT]+= s->slev; + matrix[FRONT_RIGHT][SIDE_RIGHT]+= s->slev; + }else if(s->out_ch_layout & AV_CH_FRONT_CENTER){ + matrix[ FRONT_CENTER][SIDE_LEFT ]+= s->slev*sqrt(0.5); + matrix[ FRONT_CENTER][SIDE_RIGHT]+= s->slev*sqrt(0.5); + }else + av_assert0(0); + } + + if(unaccounted & AV_CH_FRONT_LEFT_OF_CENTER){ + if(s->out_ch_layout & AV_CH_FRONT_LEFT){ + matrix[ FRONT_LEFT][ FRONT_LEFT_OF_CENTER]+= 1.0; + matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER]+= 1.0; + }else if(s->out_ch_layout & AV_CH_FRONT_CENTER){ + matrix[ FRONT_CENTER][ FRONT_LEFT_OF_CENTER]+= sqrt(0.5); + matrix[ FRONT_CENTER][FRONT_RIGHT_OF_CENTER]+= sqrt(0.5); + }else + av_assert0(0); + } + + //FIXME quantize for integeres + for(out_i=i=0; i<64; i++){ + double sum=0; + int in_i=0; + int ch_in=0; + for(j=0; j<64; j++){ + s->matrix[out_i][in_i]= matrix[i][j]; + if(matrix[i][j]){ + s->matrix_ch[out_i][++ch_in]= in_i; + sum += fabs(matrix[i][j]); + } + if(s->in_ch_layout & (1ULL<<j)) + in_i++; + } + s->matrix_ch[out_i][0]= ch_in; + maxcoef= FFMAX(maxcoef, sum); + if(s->out_ch_layout & (1ULL<<i)) + out_i++; + } + if(( s->out_sample_fmt < AV_SAMPLE_FMT_FLT + || s->int_sample_fmt < AV_SAMPLE_FMT_FLT) && maxcoef > 1.0){ + for(i=0; i<SWR_CH_MAX; i++) + for(j=0; j<SWR_CH_MAX; j++) + s->matrix[i][j] /= maxcoef; + } + for(i=0; i<av_get_channel_layout_nb_channels(s->out_ch_layout); i++){ + for(j=0; j<av_get_channel_layout_nb_channels(s->in_ch_layout); j++){ + av_log(NULL, AV_LOG_ERROR, "%f ", s->matrix[i][j]); + } + av_log(NULL, AV_LOG_ERROR, "\n"); + } + return 0; +} + +int swr_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy){ + int out_i, in_i, i, j; + +av_assert0(out->ch_count == av_get_channel_layout_nb_channels(s->out_ch_layout)); +av_assert0(in ->ch_count == av_get_channel_layout_nb_channels(s-> in_ch_layout)); + + for(out_i=0; out_i<out->ch_count; out_i++){ + switch(s->matrix_ch[out_i][0]){ + case 1: + in_i= s->matrix_ch[out_i][1]; + if(mustcopy || s->matrix[out_i][in_i]!=1.0){ + if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){ + copy_float(out->ch[out_i], in->ch[in_i], s->matrix[out_i][in_i], len); + }else + copy_s16 (out->ch[out_i], in->ch[in_i], s->matrix[out_i][in_i], len); + }else{ + out->ch[out_i]= in->ch[in_i]; + } + break; + case 2: + if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){ + sum2_float(out->ch[out_i], in->ch[ s->matrix_ch[out_i][1] ], in->ch[ s->matrix_ch[out_i][2] ], + s->matrix[out_i][ s->matrix_ch[out_i][1] ], s->matrix[out_i][ s->matrix_ch[out_i][2] ], + len); + }else{ + sum2_s16 (out->ch[out_i], in->ch[ s->matrix_ch[out_i][1] ], in->ch[ s->matrix_ch[out_i][2] ], + s->matrix[out_i][ s->matrix_ch[out_i][1] ], s->matrix[out_i][ s->matrix_ch[out_i][2] ], + len); + } + break; + default: + if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){ + for(i=0; i<len; i++){ + float v=0; + for(j=0; j<s->matrix_ch[out_i][0]; j++){ + in_i= s->matrix_ch[out_i][1+j]; + v+= ((float*)in->ch[in_i])[i] * s->matrix[out_i][in_i]; + } + ((float*)out->ch[out_i])[i]= v; + } + }else{ + for(i=0; i<len; i++){ + int v=0; + for(j=0; j<s->matrix_ch[out_i][0]; j++){ + in_i= s->matrix_ch[out_i][1+j]; + v+= ((int16_t*)in->ch[in_i])[i] * s->matrix[out_i][in_i]; //FIXME use int16 coeffs + } + ((int16_t*)out->ch[out_i])[i]= v; + } + } + } + } + return 0; +} diff --git a/libswresample/rematrix_template.c b/libswresample/rematrix_template.c new file mode 100644 index 0000000..5d5aef2 --- /dev/null +++ b/libswresample/rematrix_template.c @@ -0,0 +1,38 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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 + */ + + +static void RENAME(sum2)(SAMPLE *out, const SAMPLE *in1, const SAMPLE *in2, float coeff1, float coeff2, int len){ + int i; + + for(i=0; i<len; i++) + out[i] = coeff1*in1[i] + coeff2*in2[i]; //FIXME better int16 +} + +static void RENAME(copy)(SAMPLE *out, const SAMPLE *in, float coeff, int len){ + if(coeff == 1.0){ + memcpy(out, in, sizeof(SAMPLE)*len); + }else{ + int i; + for(i=0; i<len; i++) + out[i] = coeff*in[i]; //FIXME better int16 + } +} + diff --git a/libswresample/resample2.c b/libswresample/resample2.c new file mode 100644 index 0000000..8623877 --- /dev/null +++ b/libswresample/resample2.c @@ -0,0 +1,352 @@ +/* + * audio resampling + * Copyright (c) 2004 Michael Niedermayer <michaelni(a)gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * audio resampling + * @author Michael Niedermayer <michaelni(a)gmx.at> + */ + +#include "libavutil/log.h" +#include "swresample_internal.h" + +#ifndef CONFIG_RESAMPLE_HP +#define FILTER_SHIFT 15 + +#define FELEM int16_t +#define FELEM2 int32_t +#define FELEML int64_t +#define FELEM_MAX INT16_MAX +#define FELEM_MIN INT16_MIN +#define WINDOW_TYPE 9 +#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) +#define FILTER_SHIFT 30 + +#define FELEM int32_t +#define FELEM2 int64_t +#define FELEML int64_t +#define FELEM_MAX INT32_MAX +#define FELEM_MIN INT32_MIN +#define WINDOW_TYPE 12 +#else +#define FILTER_SHIFT 0 + +#define FELEM double +#define FELEM2 double +#define FELEML double +#define WINDOW_TYPE 24 +#endif + + +typedef struct AVResampleContext{ + const AVClass *av_class; + FELEM *filter_bank; + int filter_length; + int ideal_dst_incr; + int dst_incr; + int index; + int frac; + int src_incr; + int compensation_distance; + int phase_shift; + int phase_mask; + int linear; + double factor; +}AVResampleContext; + +/** + * 0th order modified bessel function of the first kind. + */ +static double bessel(double x){ + double v=1; + double lastv=0; + double t=1; + int i; + static const double inv[100]={ + 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10), + 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20), + 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30), + 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40), + 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50), + 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60), + 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70), + 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80), + 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90), + 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000) + }; + + x= x*x/4; + for(i=0; v != lastv; i++){ + lastv=v; + t *= x*inv[i]; + v += t; + } + return v; +} + +/** + * builds a polyphase filterbank. + * @param factor resampling factor + * @param scale wanted sum of coefficients for each filter + * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 + * @return 0 on success, negative on error + */ +static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ + int ph, i; + double x, y, w; + double *tab = av_malloc(tap_count * sizeof(*tab)); + const int center= (tap_count-1)/2; + + if (!tab) + return AVERROR(ENOMEM); + + /* if upsampling, only need to interpolate, no filter */ + if (factor > 1.0) + factor = 1.0; + + for(ph=0;ph<phase_count;ph++) { + double norm = 0; + for(i=0;i<tap_count;i++) { + x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; + if (x == 0) y = 1.0; + else y = sin(x) / x; + switch(type){ + case 0:{ + const float d= -0.5; //first order derivative = -0.5 + x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); + if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); + else y= d*(-4 + 8*x - 5*x*x + x*x*x); + break;} + case 1: + w = 2.0*x / (factor*tap_count) + M_PI; + y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); + break; + default: + w = 2.0*x / (factor*tap_count*M_PI); + y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); + break; + } + + tab[i] = y; + norm += y; + } + + /* normalize so that an uniform color remains the same */ + for(i=0;i<tap_count;i++) { +#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE + filter[ph * tap_count + i] = tab[i] / norm; +#else + filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); +#endif + } + } +#if 0 + { +#define LEN 1024 + int j,k; + double sine[LEN + tap_count]; + double filtered[LEN]; + double maxff=-2, minff=2, maxsf=-2, minsf=2; + for(i=0; i<LEN; i++){ + double ss=0, sf=0, ff=0; + for(j=0; j<LEN+tap_count; j++) + sine[j]= cos(i*j*M_PI/LEN); + for(j=0; j<LEN; j++){ + double sum=0; + ph=0; + for(k=0; k<tap_count; k++) + sum += filter[ph * tap_count + k] * sine[k+j]; + filtered[j]= sum / (1<<FILTER_SHIFT); + ss+= sine[j + center] * sine[j + center]; + ff+= filtered[j] * filtered[j]; + sf+= sine[j + center] * filtered[j]; + } + ss= sqrt(2*ss/LEN); + ff= sqrt(2*ff/LEN); + sf= 2*sf/LEN; + maxff= FFMAX(maxff, ff); + minff= FFMIN(minff, ff); + maxsf= FFMAX(maxsf, sf); + minsf= FFMIN(minsf, sf); + if(i%11==0){ + av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); + minff=minsf= 2; + maxff=maxsf= -2; + } + } + } +#endif + + av_free(tab); + return 0; +} + +AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ + double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); + int phase_count= 1<<phase_shift; + + if(!c || c->phase_shift!=phase_shift || c->linear!=linear || c->factor != factor + || c->filter_length!=FFMAX((int)ceil(filter_size/factor), 1)){ + c= av_mallocz(sizeof(AVResampleContext)); + if (!c) + return NULL; + + c->phase_shift= phase_shift; + c->phase_mask= phase_count-1; + c->linear= linear; + c->factor= factor; + + c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); + c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); + if (!c->filter_bank) + goto error; + if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) + goto error; + memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); + c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; + } + + c->compensation_distance= 0; + c->src_incr= out_rate; + c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; + c->index= -phase_count*((c->filter_length-1)/2); + c->frac= 0; + + return c; +error: + av_free(c->filter_bank); + av_free(c); + return NULL; +} + +void swr_resample_free(AVResampleContext **c){ + if(!*c) + return; + av_freep(&(*c)->filter_bank); + av_freep(c); +} + +void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){ + AVResampleContext *c= s->resample; +// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; + c->compensation_distance= compensation_distance; + c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; +} + +int swr_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ + int dst_index, i; + int index= c->index; + int frac= c->frac; + int dst_incr_frac= c->dst_incr % c->src_incr; + int dst_incr= c->dst_incr / c->src_incr; + int compensation_distance= c->compensation_distance; + + if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ + int64_t index2= ((int64_t)index)<<32; + int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; + dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); + + for(dst_index=0; dst_index < dst_size; dst_index++){ + dst[dst_index] = src[index2>>32]; + index2 += incr; + } + frac += dst_index * dst_incr_frac; + index += dst_index * dst_incr; + index += frac / c->src_incr; + frac %= c->src_incr; + }else{ + for(dst_index=0; dst_index < dst_size; dst_index++){ + FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); + int sample_index= index >> c->phase_shift; + FELEM2 val=0; + + if(sample_index < 0){ + for(i=0; i<c->filter_length; i++) + val += src[FFABS(sample_index + i) % src_size] * filter[i]; + }else if(sample_index + c->filter_length > src_size){ + break; + }else if(c->linear){ + FELEM2 v2=0; + for(i=0; i<c->filter_length; i++){ + val += src[sample_index + i] * (FELEM2)filter[i]; + v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; + } + val+=(v2-val)*(FELEML)frac / c->src_incr; + }else{ + for(i=0; i<c->filter_length; i++){ + val += src[sample_index + i] * (FELEM2)filter[i]; + } + } + +#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE + dst[dst_index] = av_clip_int16(lrintf(val)); +#else + val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; + dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; +#endif + + frac += dst_incr_frac; + index += dst_incr; + if(frac >= c->src_incr){ + frac -= c->src_incr; + index++; + } + + if(dst_index + 1 == compensation_distance){ + compensation_distance= 0; + dst_incr_frac= c->ideal_dst_incr % c->src_incr; + dst_incr= c->ideal_dst_incr / c->src_incr; + } + } + } + *consumed= FFMAX(index, 0) >> c->phase_shift; + if(index>=0) index &= c->phase_mask; + + if(compensation_distance){ + compensation_distance -= dst_index; + assert(compensation_distance > 0); + } + if(update_ctx){ + c->frac= frac; + c->index= index; + c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; + c->compensation_distance= compensation_distance; + } +#if 0 + if(update_ctx && !c->compensation_distance){ +#undef rand + av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); +av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); + } +#endif + + return dst_index; +} + +int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ + int i, ret= -1; + + for(i=0; i<dst->ch_count; i++){ + ret= swr_resample(c, dst->ch[i], src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); + } + + return ret; +} diff --git a/libswresample/swresample.c b/libswresample/swresample.c new file mode 100644 index 0000000..db2d1df --- /dev/null +++ b/libswresample/swresample.c @@ -0,0 +1,432 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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" + +#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, 0}, +{"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 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_resample_free(&s->resample); + } + + av_freep(ss); +} + +static int64_t guess_layout(int ch){ + switch(ch){ + case 1: return AV_CH_LAYOUT_MONO; + case 2: return AV_CH_LAYOUT_STEREO; + case 5: return AV_CH_LAYOUT_5POINT0; + case 6: return AV_CH_LAYOUT_5POINT1; + case 7: return AV_CH_LAYOUT_7POINT0; + case 8: return AV_CH_LAYOUT_7POINT1; + default: return 0; + } +} + +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); + + //We assume AVOptions checked the various values and the defaults where allowed + 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_layout) + s-> in_ch_layout= guess_layout(s->in.ch_count); + if(!s->out_ch_layout) + s->out_ch_layout= guess_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; + + 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); +} + +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; + } + + av_assert0(in ->planar == 0); + av_assert0(out->planar == 0); + for(i=0; i<s-> in.ch_count; i++) + in ->ch[i]= in_arg[0] + i* in->bps; + for(i=0; i<s->out.ch_count; i++) + out->ch[i]= out_arg[0] + i*out->bps; + +// 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; + int ch_count= s->resample_first ? s->in.ch_count : s->out.ch_count; + + 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; +} diff --git a/libswresample/swresample.h b/libswresample/swresample.h new file mode 100644 index 0000000..05c4f6d --- /dev/null +++ b/libswresample/swresample.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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 + */ + +#ifndef SWR_H +#define SWR_H + +#include <inttypes.h> +#include "libavutil/samplefmt.h" + +#define LIBSWRESAMPLE_VERSION_MAJOR 0 +#define LIBSWRESAMPLE_VERSION_MINOR 0 +#define LIBSWRESAMPLE_VERSION_MICRO 0 + +#define SWR_CH_MAX 16 + +#define SWR_FLAG_RESAMPLE 1///< Force resampling even if equal sample rate +//TODO use int resample ? +//long term TODO can we enable this dynamically? + + +struct SwrContext; + +/** + * Allocate SwrContext. + * @see swr_init(),swr_free() + * @return NULL on error + */ +struct SwrContext *swr_alloc(void); + +/** + * Initialize context after user parameters have been set. + * @return negativo n error + */ +int swr_init(struct SwrContext *s); + +/** + * Allocate SwrContext. + * @see swr_init(),swr_free() + * @return NULL on error + */ +struct 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); + +/** + * Free the given SwrContext. + * And set the pointer to NULL + */ +void swr_free(struct SwrContext **s); + +/** + * Convert audio. + * @param in_count Number of input samples available in one channel. + * @param out_count Amount of space available for output in samples per channel. + * @return number of samples output per channel + */ +int swr_convert(struct SwrContext *s, uint8_t *out[SWR_CH_MAX], int out_count, + const uint8_t *in [SWR_CH_MAX], int in_count); + +void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance); + +#endif diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h new file mode 100644 index 0000000..17a93d8 --- /dev/null +++ b/libswresample/swresample_internal.h @@ -0,0 +1,77 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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 + */ + +#ifndef SWR_INTERNAL_H +#define SWR_INTERNAL_H + +#include "swresample.h" + +typedef struct AudioData{ + uint8_t *ch[SWR_CH_MAX]; + uint8_t *data; + int ch_count; + int bps; + int count; + int planar; +} AudioData; + +typedef struct SwrContext { //FIXME find unused fields + AVClass *av_class; + int log_level_offset; + void *log_ctx; + enum AVSampleFormat in_sample_fmt; + enum AVSampleFormat int_sample_fmt; ///<AV_SAMPLE_FMT_FLT OR AV_SAMPLE_FMT_S16 + enum AVSampleFormat out_sample_fmt; + int64_t in_ch_layout; + int64_t out_ch_layout; + int in_sample_rate; + int out_sample_rate; + int flags; + float slev, clev; + + //below are private + int int_bps; + int resample_first; + int rematrix; ///< flag to indicate if rematrixing is used + + AudioData in, postin, midbuf, preout, out, in_buffer; + int in_buffer_index; + int in_buffer_count; + int resample_in_constraint; + + struct AVAudioConvert *in_convert; + struct AVAudioConvert *out_convert; + struct AVResampleContext *resample; + + float matrix[SWR_CH_MAX][SWR_CH_MAX]; + uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; + + //TODO callbacks for asm optims +}SwrContext; + +struct AVResampleContext *swr_resample_init(struct AVResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff); +void swr_resample_free(struct AVResampleContext **c); +int swr_multiple_resample(struct AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); +void swr_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance); +int swr_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx); + +int swr_rematrix_init(SwrContext *s); +int swr_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy); +#endif diff --git a/libswresample/swresample_test.c b/libswresample/swresample_test.c new file mode 100644 index 0000000..b634151 --- /dev/null +++ b/libswresample/swresample_test.c @@ -0,0 +1,149 @@ +/* + * Copyright (C) 2011 Michael Niedermayer (michaelni(a)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 General Public License as published by + * the Free Software Foundation; either version 2 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 General Public License for more details. + * + * You should have received a copy of the GNU 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/avassert.h" +#include "libavutil/common.h" +#include "libavutil/audioconvert.h" +#include "swresample.h" +#undef fprintf + +#define SAMPLES 1000 + +#define ASSERT_LEVEL 2 + +static double get(const void *p, int index, enum AVSampleFormat f){ + switch(f){ + case AV_SAMPLE_FMT_U8 : return ((const uint8_t*)p)[index]/255.0*2-1.0; + case AV_SAMPLE_FMT_S16: return ((const int16_t*)p)[index]/32767.0; + case AV_SAMPLE_FMT_S32: return ((const int32_t*)p)[index]/2147483647.0; + case AV_SAMPLE_FMT_FLT: return ((const float *)p)[index]; + case AV_SAMPLE_FMT_DBL: return ((const double *)p)[index]; + default: av_assert2(0); + } +} + +static void set(void *p, int index, enum AVSampleFormat f, double v){ + switch(f){ + case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= (v+1.0)*255.0/2; break; + case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= v*32767; break; + case AV_SAMPLE_FMT_S32: ((int32_t*)p)[index]= v*2147483647; break; + case AV_SAMPLE_FMT_FLT: ((float *)p)[index]= v; break; + case AV_SAMPLE_FMT_DBL: ((double *)p)[index]= v; break; + default: av_assert2(0); + } +} + +uint64_t layouts[]={ +AV_CH_LAYOUT_MONO , +AV_CH_LAYOUT_STEREO , +AV_CH_LAYOUT_2_1 , +AV_CH_LAYOUT_SURROUND , +AV_CH_LAYOUT_4POINT0 , +AV_CH_LAYOUT_2_2 , +AV_CH_LAYOUT_QUAD , +AV_CH_LAYOUT_5POINT0 , +AV_CH_LAYOUT_5POINT1 , +AV_CH_LAYOUT_5POINT0_BACK , +AV_CH_LAYOUT_5POINT1_BACK , +AV_CH_LAYOUT_7POINT0 , +AV_CH_LAYOUT_7POINT1 , +AV_CH_LAYOUT_7POINT1_WIDE , +0 +}; + +int main(int argc, char **argv){ + int in_sample_rate, out_sample_rate, ch ,i, in_ch_layout_index, out_ch_layout_index, osr; + uint64_t in_ch_layout, out_ch_layout; + enum AVSampleFormat in_sample_fmt, out_sample_fmt; + int sample_rates[]={8000,11025,16000,22050,32000}; + uint8_t array_in[SAMPLES*8*8]; + uint8_t array_mid[SAMPLES*8*8*3]; + uint8_t array_out[SAMPLES*8*8+100]; + struct SwrContext * forw_ctx= NULL; + struct SwrContext *backw_ctx= NULL; + + in_sample_rate=16000; + for(osr=0; osr<5; osr++){ + out_sample_rate= sample_rates[osr]; + for(in_sample_fmt= AV_SAMPLE_FMT_U8; in_sample_fmt<=AV_SAMPLE_FMT_DBL; in_sample_fmt++){ + for(out_sample_fmt= AV_SAMPLE_FMT_U8; out_sample_fmt<=AV_SAMPLE_FMT_DBL; out_sample_fmt++){ + for(in_ch_layout_index=0; layouts[in_ch_layout_index]; in_ch_layout_index++){ + in_ch_layout= layouts[in_ch_layout_index]; + int in_ch_count= av_get_channel_layout_nb_channels(in_ch_layout); + for(out_ch_layout_index=0; layouts[out_ch_layout_index]; out_ch_layout_index++){ + int out_count, mid_count; + out_ch_layout= layouts[out_ch_layout_index]; + int out_ch_count= av_get_channel_layout_nb_channels(out_ch_layout); + fprintf(stderr, "ch %d->%d, rate:%5d->%5d, fmt:%s->%s", + in_ch_count, out_ch_count, + in_sample_rate, out_sample_rate, + av_get_sample_fmt_name(in_sample_fmt), av_get_sample_fmt_name(out_sample_fmt)); + forw_ctx = swr_alloc2(forw_ctx, out_ch_layout, out_sample_fmt, out_sample_rate, + in_ch_layout, in_sample_fmt, in_sample_rate, 0, 0); + backw_ctx = swr_alloc2(backw_ctx,in_ch_layout, in_sample_fmt, in_sample_rate, + out_ch_layout, out_sample_fmt, out_sample_rate, 0, 0); + if(swr_init( forw_ctx) < 0) + fprintf(stderr, "swr_init(->) failed\n"); + if(swr_init(backw_ctx) < 0) + fprintf(stderr, "swr_init(<-) failed\n"); + if(!forw_ctx) + fprintf(stderr, "Failed to init forw_cts\n"); + if(!backw_ctx) + fprintf(stderr, "Failed to init backw_ctx\n"); + //FIXME test planar + for(ch=0; ch<in_ch_count; ch++){ + for(i=0; i<SAMPLES; i++) + set(array_in, ch + i*in_ch_count, in_sample_fmt, sin(i*i*3/SAMPLES)); + } + mid_count= swr_convert(forw_ctx, ( uint8_t*[]){array_mid}, 3*SAMPLES, + (const uint8_t*[]){array_in }, SAMPLES); + out_count= swr_convert(backw_ctx,( uint8_t*[]){array_out}, 3*SAMPLES, + (const uint8_t*[]){array_mid}, mid_count); + for(ch=0; ch<in_ch_count; ch++){ + double sse, x, maxdiff=0; + double sum_a= 0; + double sum_b= 0; + double sum_aa= 0; + double sum_bb= 0; + double sum_ab= 0; + for(i=0; i<SAMPLES; i++){ + double a= get(array_in , ch + i*in_ch_count, in_sample_fmt); + double b= get(array_out, ch + i*in_ch_count, in_sample_fmt); + sum_a += a; + sum_b += b; + sum_aa+= a*a; + sum_bb+= b*b; + sum_ab+= a*b; + maxdiff= FFMAX(maxdiff, FFABS(a-b)); + } + x = sum_ab/sum_bb; + sse= sum_aa + sum_bb*x*x - 2*x*sum_ab; + + fprintf(stderr, "[%f %f %f] len:%5d\n", sqrt(sse/SAMPLES), x, maxdiff, out_count); + } + fprintf(stderr, "\n"); + } + } + } + } + } + + return 0; +} diff --git a/tests/ref/acodec/g726 b/tests/ref/acodec/g726 index d2dbe50..32e1a48 100644 --- a/tests/ref/acodec/g726 +++ b/tests/ref/acodec/g726 @@ -1,4 +1,4 @@ -fd090ddf05cc3401cc75c4a5ace1d05a *./tests/data/acodec/g726.wav +a76fc937faac62c5de057cd69191732a *./tests/data/acodec/g726.wav 24052 ./tests/data/acodec/g726.wav -74abea06027375111eeac1b2f8c7d3af *./tests/data/g726.acodec.out.wav -stddev: 8554.55 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400 +124de13e6cb5af64ea8758aa49feb7fc *./tests/data/g726.acodec.out.wav +stddev: 8554.23 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400 diff --git a/tests/ref/acodec/pcm b/tests/ref/acodec/pcm index fc9dd8f..5c828b1 100644 --- a/tests/ref/acodec/pcm +++ b/tests/ref/acodec/pcm @@ -66,7 +66,7 @@ stddev: 0.00 PSNR:999.99 MAXDIFF: 0 bytes: 1058400/ 1058400 529256 ./tests/data/acodec/pcm_zork.wav 864c8c866ac25642c29a13b122c70709 *./tests/data/pcm.acodec.out.wav stddev: 633.11 PSNR: 40.30 MAXDIFF:32768 bytes: 1058400/ 1058400 -8168a5c1343553ef027541830f2cb879 *./tests/data/acodec/pcm_s24daud.302 +1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302 10368730 ./tests/data/acodec/pcm_s24daud.302 -f552afadfdfcd6348a07095da6382de5 *./tests/data/pcm.acodec.out.wav -stddev: 9416.28 PSNR: 16.85 MAXDIFF:42744 bytes: 6911796/ 1058400 +4708f86529c594e29404603c64bb208c *./tests/data/pcm.acodec.out.wav +stddev: 8967.92 PSNR: 17.28 MAXDIFF:42548 bytes: 6911796/ 1058400 diff --git a/tests/ref/lavf/dv_fmt b/tests/ref/lavf/dv_fmt index f39ff39..7450720 100644 --- a/tests/ref/lavf/dv_fmt +++ b/tests/ref/lavf/dv_fmt @@ -1,3 +1,3 @@ -188f804bd2d10cd436c8a7b111bdcd2a *./tests/data/lavf/lavf.dv +6e716216d5f9e3819db8eb8796de9129 *./tests/data/lavf/lavf.dv 3600000 ./tests/data/lavf/lavf.dv -./tests/data/lavf/lavf.dv CRC=0x02c0af30 +./tests/data/lavf/lavf.dv CRC=0x92d1e3f0 diff --git a/tests/ref/lavf/mxf_d10 b/tests/ref/lavf/mxf_d10 index 2582022..1e0ee2e 100644 --- a/tests/ref/lavf/mxf_d10 +++ b/tests/ref/lavf/mxf_d10 @@ -1,3 +1,3 @@ -b3174e2db508564c1cce0b5e3c1bc1bd *./tests/data/lavf/lavf.mxf_d10 +8eb67301f72f2b5860fafab422b920ad *./tests/data/lavf/lavf.mxf_d10 5330989 ./tests/data/lavf/lavf.mxf_d10 -./tests/data/lavf/lavf.mxf_d10 CRC=0xc3f4f92e +./tests/data/lavf/lavf.mxf_d10 CRC=0x96c02dfd

3 2

lavf/utils: ignore subtitle timestamps in timings calculation
by git＠videolan.org 05 Oct '11

05 Oct '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Fri Sep 30 17:01:51 2011 +0200| [78972a69e3a2ee940ab600f02bf09e615305ae0d] | committer: Michael Niedermayer lavf/utils: ignore subtitle timestamps in timings calculation Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=78972a69e3a2ee940… --- libavformat/utils.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/libavformat/utils.c b/libavformat/utils.c index 7c80959..bb926d5 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -1898,7 +1898,7 @@ static void update_stream_timings(AVFormatContext *ic) st = ic->streams[i]; if (st->start_time != AV_NOPTS_VALUE && st->time_base.den) { start_time1= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q); - if (st->codec->codec_id == CODEC_ID_DVB_TELETEXT) { + if (st->codec->codec_id == CODEC_ID_DVB_TELETEXT || st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) { if (start_time1 < start_time_text) start_time_text = start_time1; } else

3 2

celp_math: cleanup ff_dot_product()
by git＠videolan.org 01 Oct '11

01 Oct '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Thu Sep 29 21:21:26 2011 +0200| [11512367d3c6f67b093b54c532328763c9b815e2] | committer: Michael Niedermayer celp_math: cleanup ff_dot_product() based on code & idea by vitor Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=11512367d3c6f67b0… --- libavcodec/celp_math.c | 12 ++++++------ libavcodec/celp_math.h | 3 +-- libavcodec/g723_1.c | 18 +++++++++--------- 3 files changed, 16 insertions(+), 17 deletions(-) diff --git a/libavcodec/celp_math.c b/libavcodec/celp_math.c index b78edd1..a00bb39 100644 --- a/libavcodec/celp_math.c +++ b/libavcodec/celp_math.c @@ -197,14 +197,14 @@ int ff_log2(uint32_t value) return (power_int << 15) + value; } -int ff_dot_product(const int16_t *a, const int16_t *b, int length, int shift) +int ff_dot_product(const int16_t *a, const int16_t *b, int length) { - int i, sum = 0; + int i; + int64_t sum = 0; + + for (i = 0; i < length; i++) + sum += MUL16(a[i], b[i]); - for (i = 0; i < length; i++) { - int64_t prod = av_clipl_int32(MUL64(a[i], b[i]) << shift); - sum = av_clipl_int32(sum + prod); - } return sum; } diff --git a/libavcodec/celp_math.h b/libavcodec/celp_math.h index 476e668..4b08c62 100644 --- a/libavcodec/celp_math.h +++ b/libavcodec/celp_math.h @@ -68,11 +68,10 @@ static inline int bidir_sal(int value, int offset) * @param a input data array * @param b input data array * @param length number of elements - * @param shift the result is scaled by 2^shift * * @return dot product = sum of elementwise products */ -int ff_dot_product(const int16_t *a, const int16_t *b, int length, int shift); +int ff_dot_product(const int16_t *a, const int16_t *b, int length); /** * returns the dot product. diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c index da4f0d2..e0e866e 100644 --- a/libavcodec/g723_1.c +++ b/libavcodec/g723_1.c @@ -553,7 +553,7 @@ static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, /* Calculate adaptive vector */ cb_ptr += subfrm.ad_cb_gain * 20; for (i = 0; i < SUBFRAME_LEN; i++) { - sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER, 1); + sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER)<<1; vector[i] = av_clipl_int32((sum << 1) + (1 << 15)) >> 16; } } @@ -579,7 +579,7 @@ static int autocorr_max(G723_1_Context *p, int offset, int *ccr_max, limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3); for (i = pitch_lag - 3; i <= limit; i++) { - ccr = ff_dot_product(buf, buf + dir * i, length, 1); + ccr = ff_dot_product(buf, buf + dir * i, length)<<1; if (ccr > *ccr_max) { *ccr_max = ccr; @@ -678,17 +678,17 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, return; /* Compute target energy */ - energy[0] = ff_dot_product(buf, buf, SUBFRAME_LEN, 1); + energy[0] = ff_dot_product(buf, buf, SUBFRAME_LEN)<<1; /* Compute forward residual energy */ if (fwd_lag) energy[2] = ff_dot_product(buf + fwd_lag, buf + fwd_lag, - SUBFRAME_LEN, 1); + SUBFRAME_LEN)<<1; /* Compute backward residual energy */ if (back_lag) energy[4] = ff_dot_product(buf - back_lag, buf - back_lag, - SUBFRAME_LEN, 1); + SUBFRAME_LEN)<<1; /* Normalize and shorten */ temp1 = 0; @@ -749,7 +749,7 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag, ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16; /* Compute target energy */ - tgt_eng = ff_dot_product(buf, buf, SUBFRAME_LEN * 2, 1); + tgt_eng = ff_dot_product(buf, buf, SUBFRAME_LEN * 2)<<1; *exc_eng = av_clipl_int32(tgt_eng + (1 << 15)) >> 16; if (ccr <= 0) @@ -757,7 +757,7 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag, /* Compute best energy */ best_eng = ff_dot_product(buf - index, buf - index, - SUBFRAME_LEN * 2, 1); + SUBFRAME_LEN * 2)<<1; best_eng = av_clipl_int32((int64_t)best_eng + (1 << 15)) >> 16; temp = best_eng * *exc_eng >> 3; @@ -911,9 +911,9 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf) /* Compute auto correlation coefficients */ auto_corr[0] = ff_dot_product(temp_vector, temp_vector + 1, - SUBFRAME_LEN - 1, 1); + SUBFRAME_LEN - 1)<<1; auto_corr[1] = ff_dot_product(temp_vector, temp_vector, - SUBFRAME_LEN, 1); + SUBFRAME_LEN)<<1; /* Compute reflection coefficient */ temp = auto_corr[1] >> 16;

3 4

Add G.723.1 decoder data tables
by git＠videolan.org 01 Oct '11

01 Oct '11

ffmpeg | branch: master | Mohamed Naufal Basheer <naufal11(a)gmail.com> | Thu Mar 17 23:56:49 2011 +0100| [da43980fa5a3d4a7240bcb7ae0a458bef3379fa9] | committer: Michael Niedermayer Add G.723.1 decoder data tables > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=da43980fa5a3d4a72… --- libavcodec/g723_1_data.h | 1245 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 1245 insertions(+), 0 deletions(-) diff --git a/libavcodec/g723_1_data.h b/libavcodec/g723_1_data.h new file mode 100644 index 0000000..3b180ef --- /dev/null +++ b/libavcodec/g723_1_data.h @@ -0,0 +1,1245 @@ +/* + * G723.1 compatible decoder data tables. + * Copyright (c) 2006 Benjamin Larsson + * Copyright (c) 2010 Mohamed Naufal Basheer + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * G723.1 compatible decoder data tables + */ + +#define SUBFRAMES 4 +#define SUBFRAME_LEN 60 +#define FRAME_LEN (SUBFRAME_LEN << 2) +#define LPC_ORDER 10 +#define LSP_BANDS 3 +#define LSP_CB_SIZE 256 +#define PITCH_MIN 18 +#define PITCH_MAX (PITCH_MIN + 127) +#define PITCH_ORDER 5 +#define GRID_SIZE 2 +#define PULSE_MAX 6 +#define GAIN_LEVELS 24 +#define COS_TBL_SIZE 512 + +/** + * G723.1 frame types + */ +typedef enum { + ActiveFrame, ///< Active speech + SIDFrame, ///< Silence Insertion Descriptor frame + UntransmittedFrame +} FrameType; + +static const uint8_t frame_size[4] = {24, 20, 4, 1}; + +typedef enum { + Rate6k3, + Rate5k3 +} Rate; + +/** + * G723.1 unpacked data subframe + */ +typedef struct { + int ad_cb_lag; ///< adaptive codebook lag + int ad_cb_gain; + int dirac_train; + int pulse_sign; + int grid_index; + int amp_index; + int pulse_pos; +} G723_1_Subframe; + +/** + * Pitch postfilter parameters + */ +typedef struct { + int index; ///< postfilter backward/forward lag + int16_t opt_gain; ///< optimal gain + int16_t sc_gain; ///< scaling gain +} PPFParam; + +/** + * Postfilter gain weighting factors scaled by 2^15 + */ +static const int16_t ppf_gain_weight[2] = {0x1800, 0x2000}; + +/** + * LSP DC component + */ +static const int16_t dc_lsp[LPC_ORDER] = { + 0x0c3b, + 0x1271, + 0x1e0a, + 0x2a36, + 0x3630, + 0x406f, + 0x4d28, + 0x56f4, + 0x638c, + 0x6c46 +}; + +/** + * Cosine table scaled by 2^14 + */ +static const int16_t cos_tab[COS_TBL_SIZE] = { + 16384, 16383, 16379, 16373, 16364, 16353, 16340, 16324, + 16305, 16284, 16261, 16235, 16207, 16176, 16143, 16107, + 16069, 16029, 15986, 15941, 15893, 15843, 15791, 15736, + 15679, 15619, 15557, 15493, 15426, 15357, 15286, 15213, + 15137, 15059, 14978, 14896, 14811, 14724, 14635, 14543, + 14449, 14354, 14256, 14155, 14053, 13949, 13842, 13733, + 13623, 13510, 13395, 13279, 13160, 13039, 12916, 12792, + 12665, 12537, 12406, 12274, 12140, 12004, 11866, 11727, + 11585, 11442, 11297, 11151, 11003, 10853, 10702, 10549, + 10394, 10238, 10080, 9921, 9760, 9598, 9434, 9269, + 9102, 8935, 8765, 8595, 8423, 8250, 8076, 7900, + 7723, 7545, 7366, 7186, 7005, 6823, 6639, 6455, + 6270, 6084, 5897, 5708, 5520, 5330, 5139, 4948, + 4756, 4563, 4370, 4176, 3981, 3786, 3590, 3393, + 3196, 2999, 2801, 2603, 2404, 2205, 2006, 1806, + 1606, 1406, 1205, 1005, 804, 603, 402, 201, + 0, -201, -402, -603, -804, -1005, -1205, -1406, + -1606, -1806, -2006, -2205, -2404, -2603, -2801, -2999, + -3196, -3393, -3590, -3786, -3981, -4176, -4370, -4563, + -4756, -4948, -5139, -5330, -5520, -5708, -5897, -6084, + -6270, -6455, -6639, -6823, -7005, -7186, -7366, -7545, + -7723, -7900, -8076, -8250, -8423, -8595, -8765, -8935, + -9102, -9269, -9434, -9598, -9760, -9921, -10080, -10238, + -10394, -10549, -10702, -10853, -11003, -11151, -11297, -11442, + -11585, -11727, -11866, -12004, -12140, -12274, -12406, -12537, + -12665, -12792, -12916, -13039, -13160, -13279, -13395, -13510, + -13623, -13733, -13842, -13949, -14053, -14155, -14256, -14354, + -14449, -14543, -14635, -14724, -14811, -14896, -14978, -15059, + -15137, -15213, -15286, -15357, -15426, -15493, -15557, -15619, + -15679, -15736, -15791, -15843, -15893, -15941, -15986, -16029, + -16069, -16107, -16143, -16176, -16207, -16235, -16261, -16284, + -16305, -16324, -16340, -16353, -16364, -16373, -16379, -16383, + -16384, -16383, -16379, -16373, -16364, -16353, -16340, -16324, + -16305, -16284, -16261, -16235, -16207, -16176, -16143, -16107, + -16069, -16029, -15986, -15941, -15893, -15843, -15791, -15736, + -15679, -15619, -15557, -15493, -15426, -15357, -15286, -15213, + -15137, -15059, -14978, -14896, -14811, -14724, -14635, -14543, + -14449, -14354, -14256, -14155, -14053, -13949, -13842, -13733, + -13623, -13510, -13395, -13279, -13160, -13039, -12916, -12792, + -12665, -12537, -12406, -12274, -12140, -12004, -11866, -11727, + -11585, -11442, -11297, -11151, -11003, -10853, -10702, -10549, + -10394, -10238, -10080, -9921, -9760, -9598, -9434, -9269, + -9102, -8935, -8765, -8595, -8423, -8250, -8076, -7900, + -7723, -7545, -7366, -7186, -7005, -6823, -6639, -6455, + -6270, -6084, -5897, -5708, -5520, -5330, -5139, -4948, + -4756, -4563, -4370, -4176, -3981, -3786, -3590, -3393, + -3196, -2999, -2801, -2603, -2404, -2205, -2006, -1806, + -1606, -1406, -1205, -1005, -804, -603, -402, -201, + 0, 201, 402, 603, 804, 1005, 1205, 1406, + 1606, 1806, 2006, 2205, 2404, 2603, 2801, 2999, + 3196, 3393, 3590, 3786, 3981, 4176, 4370, 4563, + 4756, 4948, 5139, 5330, 5520, 5708, 5897, 6084, + 6270, 6455, 6639, 6823, 7005, 7186, 7366, 7545, + 7723, 7900, 8076, 8250, 8423, 8595, 8765, 8935, + 9102, 9269, 9434, 9598, 9760, 9921, 10080, 10238, + 10394, 10549, 10702, 10853, 11003, 11151, 11297, 11442, + 11585, 11727, 11866, 12004, 12140, 12274, 12406, 12537, + 12665, 12792, 12916, 13039, 13160, 13279, 13395, 13510, + 13623, 13733, 13842, 13949, 14053, 14155, 14256, 14354, + 14449, 14543, 14635, 14724, 14811, 14896, 14978, 15059, + 15137, 15213, 15286, 15357, 15426, 15493, 15557, 15619, + 15679, 15736, 15791, 15843, 15893, 15941, 15986, 16029, + 16069, 16107, 16143, 16176, 16207, 16235, 16261, 16284, + 16305, 16324, 16340, 16353, 16364, 16373, 16379, 16383, +}; + +/** + * LSP VQ tables + */ +static const int16_t lsp_band0[LSP_CB_SIZE][3] = { + { 0, 0, 0}, { -270, -1372, -1032}, { -541, -1650, -1382}, + { -723, -2011, -2213}, { -941, -1122, -1942}, { -780, -1145, -2454}, + { -884, -1309, -1373}, {-1051, -1523, -1766}, {-1083, -1622, -2300}, + { -777, -1377, -2147}, { -935, -1467, -2763}, { -802, -1327, -3471}, + { -935, -1959, -3999}, { -240, -89, 222}, { -661, -257, -160}, + { -994, -466, -419}, { -188, -164, -278}, { -342, -512, -415}, + { -607, -511, -797}, { 16, 19, -716}, { 374, 425, -972}, + { -346, 245, -282}, { -265, 506, -754}, { -620, -147, 1955}, + { -742, -860, 2597}, { -150, -352, 2704}, { 305, 880, 1954}, + { 123, 731, 2766}, { -348, 765, 3327}, { 618, 221, 3258}, + { -178, -47, 4219}, { 393, 1304, 3842}, { 698, 1702, 4801}, + { 63, -584, 1229}, { -215, -732, 1704}, { 172, -335, 1909}, + { -2, 216, 1797}, { 353, 127, 2205}, {-1208, 188, 11}, + { -513, -75, -683}, { -973, 222, -646}, { -616, -843, -388}, + { -950, -1113, -359}, {-1431, -623, -705}, {-1398, -1063, -178}, + { -45, -461, 35}, { -9, -657, -216}, { 127, -1078, 95}, + { -950, -1156, 584}, {-1480, -1494, 449}, { -120, -705, 516}, + { -368, -961, 727}, { -378, -526, 973}, { -793, -614, 676}, + { -801, -755, 1287}, {-1476, -340, 1636}, { -505, -1254, 1543}, + {-1243, -1622, 1532}, { -776, -1477, -655}, {-1151, -1296, -823}, + {-1153, -1672, -1124}, {-1291, -2003, -1702}, { -622, -1283, 57}, + { -471, -1611, 509}, {-1060, -1570, -139}, { -873, -2156, -536}, + {-1716, -2021, -364}, {-2150, -3218, -1291}, {-1248, -1945, -2904}, + {-1215, -2633, -2855}, { 167, -244, 84}, { 349, -412, -217}, + { -40, -352, 632}, { 227, -529, 405}, { 68, -383, -443}, + { 167, -558, -706}, { -275, -854, -14}, { -351, -1089, -449}, + { 341, -72, -289}, { 603, -106, -474}, { 322, -219, -649}, + { 179, -317, -998}, { 450, -291, -996}, { 555, 195, -525}, + { 784, 272, -831}, { -148, -384, -849}, { 82, -536, -1357}, + { 238, -172, -1354}, { 422, -268, -1841}, { 297, -737, -2079}, + { -111, -801, -598}, { 1, -668, -984}, { -131, -818, -1299}, + { -329, -521, -1310}, { -151, -778, -1834}, { -93, -352, -1746}, + { -568, -640, -1821}, { -509, -941, -2183}, { 464, -815, -1250}, + { 79, -1133, -1597}, { -184, -1353, -2123}, { -196, -410, -2427}, + { -192, -833, -2810}, { -259, -1382, -3045}, { -217, 4, -1166}, + { -800, -325, -1219}, { -363, -830, -898}, { -661, -1134, -960}, + { -386, -980, -1501}, { -627, -1159, -1722}, { -903, -829, -855}, + { -685, -829, -1313}, {-1065, -959, -1405}, { 441, 25, -847}, + { 655, -27, -1181}, { 1159, -110, -705}, { 856, 253, -1671}, + { 415, 404, -1}, { 322, 903, -398}, { 670, 499, -292}, + { 803, 591, -610}, { 1144, 591, -814}, { 717, 183, 393}, + { 857, 381, 106}, { 609, 62, -27}, { 792, 198, -325}, + { 735, 805, 88}, { 1142, 812, 78}, { 1028, 366, -292}, + { 1309, 743, -237}, { 1615, 589, -79}, { 1010, 639, -243}, + { 999, 964, -311}, { 1500, 1137, -615}, { 988, 357, 646}, + { 1227, 667, 683}, { 1164, 1565, 894}, { 1392, 2015, 477}, + { 1138, 533, 250}, { 1437, 896, 391}, { 1765, 1118, 99}, + { 1112, 1090, 802}, { 1596, 846, 1134}, { 937, 1161, 279}, + { 1719, 1254, 683}, { 1338, 1086, 35}, { 1419, 1324, 428}, + { 1428, 1524, 40}, { 2108, 1594, 89}, { 1015, 544, 1222}, + { 1121, 925, 1263}, { 1030, 1318, 1485}, { 1295, 789, 1817}, + { 1323, 1272, 1909}, { 1724, 1237, 1803}, { 1797, 1689, 858}, + { 2149, 1367, 1301}, { 2302, 1867, 761}, { 2863, 2351, 1053}, + { 52, 163, -76}, { 230, 309, -492}, { -71, 619, 39}, + { -218, 856, 499}, { -654, 736, -207}, { -535, 1259, 155}, + { -480, 1476, 643}, { 262, 1081, 102}, { 309, 1592, -182}, + { 627, 1629, 534}, { 337, 643, 456}, { 758, 670, 713}, + { 202, 1126, 658}, { 612, 1131, 666}, { 686, 1223, 1136}, + { -131, 377, 525}, { 42, 708, 907}, { 87, 1488, 1035}, + { 432, 2117, 904}, { 137, 981, 1332}, { -447, 1014, 1136}, + { -839, 1793, 1246}, { -559, 297, 198}, { -850, 685, 446}, + {-1273, 632, 826}, { -401, -544, 173}, { -753, -793, 144}, + { -436, -9, 772}, { -115, -243, 1310}, { -670, -269, 374}, + {-1027, -13, 639}, { -887, -81, 1137}, {-1277, -455, 158}, + {-1411, -720, 736}, { 172, 88, 403}, { 386, 255, 756}, + { -500, 522, 910}, { -958, 659, 1388}, { -395, 301, 1344}, + { -356, 768, 1813}, { -613, 841, 2419}, { 445, -122, 252}, + { 629, -87, 723}, { 283, -253, 870}, { 456, -116, 1381}, + { 757, 180, 1059}, { 532, 408, 1509}, { 947, 288, 1806}, + { 1325, 994, 2524}, { 892, 1219, 3023}, { 1397, 1596, 3406}, + { 1143, 1552, 2546}, { 1850, 1433, 2710}, { -10, 134, 1002}, + { 154, 499, 1323}, { 508, 792, 1117}, { 509, 1340, 1616}, + { 762, 862, 1608}, { 787, 740, 2320}, { 794, 1727, 1283}, + { 465, 2108, 1660}, { -120, 1451, 1613}, { -386, 2016, 2169}, + { 891, 1225, 2050}, { 456, 1480, 2185}, { 1493, 1283, 1209}, + { 1397, 1636, 1518}, { 1776, 1738, 1552}, { 1572, 1698, 2141}, + { 1389, 2126, 1271}, { 1959, 2413, 1119}, { 1365, 2892, 1505}, + { 2206, 1971, 1623}, { 2076, 1950, 2280}, { 1717, 2291, 1867}, + { 2366, 2515, 1953}, { 2865, 2838, 2522}, { 2535, 3465, 2011}, + { 3381, 4127, 2638}, { 836, 2667, 2289}, { 1761, 2773, 2337}, + { 1415, 3325, 2911}, { 2354, 3138, 3126}, { 2659, 4192, 4010}, + { 1048, 1786, 1818}, { 1242, 2111, 2240}, { 1512, 2079, 2780}, + { 1573, 2491, 3138}, { 2230, 2377, 2782}, { 416, 1773, 2704}, + { 725, 2336, 3297}, { 1252, 2373, 3978}, { 2094, 2268, 3568}, + { 2011, 2712, 4528}, { 1341, 3507, 3876}, { 1216, 3919, 4922}, + { 1693, 4793, 6012} +}; + +static const int16_t lsp_band1[LSP_CB_SIZE][3] = { + { 0, 0, 0}, {-2114, -1302, 76}, {-2652, -1278, -1368}, + {-2847, -828, -349}, {-3812, -2190, -349}, {-3946, -364, -449}, + {-2725, -4492, -3607}, {-3495, -4764, -1744}, { -51, -756, 84}, + { -153, -1191, 504}, { 108, -1418, 1167}, { -835, -896, 390}, + { -569, -1702, 87}, {-1151, -1818, 933}, {-1826, -2547, 411}, + {-1842, -1818, 1451}, {-2438, -1611, 781}, {-2747, -2477, 1311}, + { -940, 1252, 477}, {-1629, 1688, 602}, {-1202, 617, 280}, + {-1737, 393, 580}, {-1528, 1077, 1199}, {-2165, -161, 1408}, + {-2504, -1087, 2371}, {-3458, -175, 1395}, {-1397, -98, -843}, + {-2252, -177, -1149}, {-1489, -726, -1283}, {-1558, -265, -1744}, + {-1867, -821, -1897}, {-2062, -1516, -2340}, {-2595, -1142, -2861}, + { 170, 46, -819}, { -193, -204, -1151}, { 326, -196, -1532}, + { 780, 329, -816}, { 201, 369, -1243}, { 650, -209, -1060}, + { 1144, -15, -1216}, { 1203, -259, -1867}, { -890, -564, -1430}, + { -638, -852, -1921}, { 177, -739, -1358}, { -261, -526, -1666}, + { 206, -407, -2255}, { 338, -526, -822}, { 421, -1095, -1009}, + { 765, -607, -1408}, { 825, -1295, -2004}, { 357, -905, -1815}, + { -58, -1248, -1588}, { -596, -1436, -2046}, { -73, -1159, -2116}, + { -115, -1382, -2581}, { -160, -1723, -1952}, { -6, -2196, -2954}, + { -649, -1705, -2603}, { -617, -1453, -3282}, { -949, -2019, -3102}, + { -812, 1544, 1937}, {-1854, 574, 2000}, {-1463, 1140, 2649}, + {-2683, 1748, 1452}, {-2486, 2241, 2523}, { 783, 1910, 1435}, + { 581, 2682, 1376}, { 236, 2197, 1885}, { -453, 2943, 2057}, + { -682, 2178, 2565}, {-1342, 3201, 3328}, { -288, -184, 262}, + { 121, -149, -183}, { 758, -412, 206}, { 1038, -204, 853}, + { 1577, -457, 700}, { 937, -640, -567}, { 1508, -528, -1024}, + { -225, -527, -427}, { -564, -1095, -332}, { -742, -353, -186}, + {-1288, -459, 84}, {-1853, -484, -274}, {-1554, -731, 825}, + {-2425, -234, 382}, {-1722, 293, -271}, {-2515, 425, -564}, + {-2599, 818, 464}, { -358, 118, -375}, { -613, 198, -874}, + { -690, 683, -324}, {-1352, 1155, -168}, {-1093, 129, -324}, + {-1184, 611, -858}, { 433, 386, -372}, { -120, 486, -634}, + { 234, 851, -631}, { 602, 128, 46}, { 1099, 410, 159}, + { 715, -145, -424}, { 1198, -85, -593}, { 1390, 367, -358}, + { 1683, 362, -964}, { 1711, 622, 45}, { 2033, 833, -383}, + { 2890, 549, -506}, { 7, 401, 52}, { 72, 811, 415}, + { 566, 668, 41}, { 467, 1218, 130}, { 68, 957, -187}, + { -25, 1649, -103}, { -661, 260, 214}, { -925, -94, 612}, + { -321, -422, 965}, { -788, -672, 1783}, { 400, -673, 779}, + { 741, -595, 1635}, { -161, 307, 657}, { -382, 836, 871}, + { -814, 400, 1223}, { 364, 606, 1247}, { 57, 75, 1571}, + { 151, 471, 2287}, { -81, 1021, 1502}, { 227, 1470, 1097}, + { 658, 1275, 1653}, { 664, 1478, 2377}, { 263, -127, 444}, + { 264, 89, 969}, { 794, 171, 576}, { 821, 186, 1226}, + { 404, 462, 517}, { 339, 918, 794}, { 1280, 1423, 196}, + { 1453, 2019, 365}, { 1615, 1481, 672}, { 2394, 1708, 508}, + { 806, 1238, 573}, { 713, 1158, 1078}, { 1285, 1436, 1232}, + { 1790, 1188, 1141}, { 765, 643, 864}, { 1032, 797, 1279}, + { 900, 563, 1827}, { 1514, 673, 2312}, { 1544, 1129, 3240}, + { 1469, 1050, 1594}, { 1945, 1318, 1988}, { 2397, 2026, 2060}, + { 3538, 2057, 2620}, { 1249, -118, 74}, { 1727, 194, 421}, + { 2078, -50, -463}, { 970, 688, -432}, { 1149, 952, -110}, + { 1254, 1275, -651}, { 1386, 929, 401}, { 1960, 1167, 232}, + { 407, -752, -243}, { 859, -1118, 172}, { -227, -860, -992}, + { -796, -1175, -1380}, { 8, -1282, -388}, { 353, -1781, -1037}, + { -732, -397, -807}, { -853, -28, -1342}, {-1229, -1207, -1959}, + {-1015, -1125, -2543}, {-1452, -1791, -2725}, {-1891, -2416, -3269}, + { -918, -1629, -783}, { -580, -2155, -698}, {-1097, -2364, -96}, + {-1387, -1513, 7}, {-1588, -2076, -664}, {-1473, -2740, -784}, + {-2378, -3149, -56}, {-2856, -2092, -169}, {-3391, -3708, 316}, + {-1176, -890, -614}, {-1944, -1061, -800}, { -299, -1517, -1000}, + { -640, -1850, -1526}, {-1454, -1536, -1233}, {-1890, -1955, -1756}, + {-1086, -1921, -2122}, { -750, -2325, -2260}, {-1325, -2413, -2673}, + {-1114, -2542, -3459}, {-1341, -2901, -3963}, {-1160, -2226, -1393}, + {-1001, -2772, -1573}, {-1594, -2641, -1978}, {-1534, -3046, -2624}, + {-2224, -2196, -675}, {-2807, -3054, -1102}, {-2008, -2840, -1186}, + {-1980, -3332, -1695}, {-1715, -3562, -505}, {-2527, -4000, -1887}, + {-2333, -2734, -2296}, {-3440, -2401, -3211}, {-2008, -3528, -3337}, + {-2247, -3291, -4510}, { -475, 949, 155}, { -149, 1365, 545}, + { -757, 1644, 1083}, { -217, 2053, 1353}, {-1433, 2301, 1462}, + { 495, 1661, 529}, { 10, 2037, 740}, { 2082, 1898, 978}, + { 2831, 2294, 911}, { 842, 793, 420}, { 1223, 1023, 863}, + { 1237, 451, 780}, { 1744, 708, 822}, { 1533, 284, 1384}, + { 2135, 609, 1538}, { 2305, 626, 540}, { 2368, 1187, 955}, + { 2586, 1255, -7}, { 3116, 1131, 726}, { 3431, 1730, 428}, + { 2734, 1648, 1307}, { 2988, 1231, 2010}, { 3523, 2024, 1488}, + { 1034, 1657, 871}, { 1206, 2163, 1036}, { 1807, 2372, 1233}, + { 1808, 1769, 1493}, { 1573, 2332, 1779}, { 1216, 1609, 1866}, + { 1480, 1898, 2513}, { 465, 2708, 2776}, { 771, 3638, 3338}, + { 1869, 2599, 2623}, { 2825, 2745, 2468}, { 2638, 2439, 1585}, + { 2094, 2970, 1308}, { 2022, 3057, 1999}, { 3428, 2912, 1816}, + { 4536, 2974, 2129}, { 1046, 2563, 2086}, { 1363, 3562, 2318}, + { 2511, 1891, 2984}, { 1866, 2306, 3986}, { 3272, 2924, 3682}, + { 3146, 3564, 2272}, { 3592, 3968, 2822}, { 2431, 3369, 3069}, + { 1931, 4709, 3090}, { 2629, 4220, 3986}, { 4639, 4056, 3664}, + { 4035, 5334, 4912} +}; + +static const int16_t lsp_band2[LSP_CB_SIZE][4] = { + { 0, 0, 0, 0}, { 601, 512, -542, 334}, + { 428, 1087, -484, -132}, { 652, 622, -391, -572}, + { 378, 799, 141, -860}, { 1040, 409, 112, -554}, + { 1123, 670, -75, -847}, { 1421, 494, -315, -1095}, + { 787, 1001, 114, -460}, { 988, 1672, 216, -681}, + { 1007, 1241, -132, -1247}, { 1073, 399, 186, -5}, + { 1262, 193, -694, -129}, { 325, 196, 51, -641}, + { 861, -59, 350, -458}, { 1261, 567, 586, -346}, + { 1532, 885, 210, -517}, { 2027, 937, 113, -792}, + { 1383, 1064, 334, 38}, { 1964, 1468, 459, 133}, + { 2062, 1186, -98, -121}, { 2577, 1445, 506, -373}, + { 2310, 1682, -2, -960}, { 2876, 1939, 765, 138}, + { 3581, 2360, 649, -414}, { 219, 176, -398, -309}, + { 434, -78, -435, -880}, { -344, 301, 265, -552}, + { -915, 470, 657, -380}, { 419, -432, -163, -453}, + { 351, -953, 8, -562}, { 789, -43, 20, -958}, + { 302, -594, -352, -1159}, { 1040, 108, -668, -924}, + { 1333, 210, -1217, -1663}, { 483, 589, -350, -1140}, + { 1003, 824, -802, -1184}, { 745, 58, -589, -1443}, + { 346, 247, -915, -1683}, { 270, 796, -720, -2043}, + { 1208, 722, -222, -193}, { 1486, 1180, -412, -672}, + { 1722, 179, -69, -521}, { 2047, 860, -666, -1410}, + { -146, 222, -281, -805}, { -189, 90, -114, -1307}, + { -152, 1086, -241, -764}, { -439, 733, -601, -1302}, + { -833, -167, -351, -601}, { -856, -422, -411, -1059}, + { -747, -355, -582, -1644}, { -837, 210, -916, -1144}, + {-1800, 32, -878, -1687}, { -48, -23, -1146, 52}, + { -350, -409, -1656, -364}, { 265, -728, -858, -577}, + { 458, -247, -1141, -997}, { 691, -407, -1988, -1161}, + { -66, -104, -705, -1249}, { -431, -93, -1191, -1844}, + { 203, -732, -1000, -1693}, { 10, -832, -1846, -1819}, + { 493, -128, -1436, -1768}, { 488, -311, -1730, -2540}, + { -653, -532, -1150, -1172}, {-1086, -289, -1706, -1533}, + { -699, -1205, -1216, -1766}, {-1032, -1481, -2074, -1523}, + { -721, -1220, -2277, -2600}, { 12, -539, -1484, -1131}, + { -40, -911, -2106, -441}, { -471, -484, -2267, -1549}, + { -141, -988, -3006, -1721}, {-1545, -2102, -583, 342}, + {-1383, -2772, -386, -13}, {-2118, -2589, -1205, 72}, + {-2147, -3231, -965, 390}, {-2949, -3300, -621, 637}, + {-3907, -4138, -865, 803}, {-1287, -845, -375, -548}, + {-1416, -1169, -487, -1277}, {-1400, -1690, -1027, -418}, + {-2018, -1909, -1188, -1260}, {-1418, -2222, -2029, -128}, + {-2067, -2998, -2693, -310}, { -950, -1028, -1538, 185}, + {-1616, -915, -2205, -549}, { 19, -821, -1145, 352}, + { 184, -1175, -1356, -627}, { -547, -1088, -1661, -911}, + { -216, -1502, -2197, -948}, { -795, -1306, -2374, -451}, + { -924, -1889, -2796, -680}, { -600, -1614, -3609, -885}, + {-2392, -2528, 319, 303}, {-2908, -2095, -310, 573}, + {-3460, -2141, 49, -113}, {-2231, -448, 675, -146}, + {-2805, -532, 1231, 479}, {-2684, -486, -200, 611}, + {-3525, -971, -198, 704}, {-3707, 173, 349, 254}, + {-4734, -1447, -34, 880}, { 777, -512, 114, -10}, + { 1250, -66, 442, -5}, { 604, 613, 452, -352}, + { 1224, 777, 675, -1014}, {-1372, -79, -1208, -238}, + {-2389, -17, -1157, -818}, {-1504, -673, -1133, -1060}, + {-1984, -799, -2005, -1973}, {-2037, -798, -1068, -105}, + {-3190, -899, -1817, -194}, { -156, -886, 394, -318}, + { -258, -1283, 551, 202}, { -536, -1729, 910, 331}, + { -847, -1109, 795, -163}, {-1171, -1128, 715, 519}, + {-1080, -1319, 1685, 668}, {-1000, -1921, 96, 211}, + {-1487, -2148, 831, 174}, {-1139, -374, 414, -4}, + {-1517, -1383, 396, -352}, {-1012, 439, -59, -967}, + {-1812, 706, -440, -1030}, {-1971, -329, -34, -827}, + {-2472, -1588, -151, -606}, {-2161, 374, -281, 76}, + {-3012, 231, -15, -690}, { 1104, 566, 721, 209}, + { 1685, 564, 383, 98}, { 1898, 750, 792, -97}, + { 556, -64, 561, -93}, { 876, 162, 913, -22}, + { 961, 675, 1296, 140}, { 756, -396, 851, 544}, + { 360, -303, 1341, 396}, { 878, -22, 1464, 863}, + { -309, -273, 642, -129}, { -686, -82, 842, 454}, + { -5, -47, 1069, 998}, { -94, 967, 1277, 298}, + { -489, 385, 1473, 746}, { -369, -717, 1333, 242}, + { 281, -993, 1726, 924}, { 464, 601, 1575, 1376}, + { -250, 206, 2339, 1175}, { -438, 377, -597, -285}, + {-1020, 787, -790, -287}, { -458, -410, 215, 295}, + { -589, -860, -121, 797}, {-1175, 122, -437, 466}, + {-1480, -121, 367, 924}, { 234, 323, 770, -555}, + { 145, 30, 996, 26}, { 66, 849, 93, -145}, + { -117, 1261, 474, -399}, {-1495, 1051, 218, -506}, + {-1390, 694, 994, 88}, { 616, 7, 78, 304}, + { 1060, 52, -62, 835}, { 833, 454, 649, 1359}, + { -770, 464, 47, 93}, { -574, 1199, -39, 379}, + { 114, -98, 488, 485}, { 727, 244, 606, 696}, + { -76, 455, 671, 546}, { -565, -13, 145, 819}, + { -376, 569, 448, 1128}, { 218, 122, 265, 1167}, + { 230, 738, 932, 1003}, { 138, 477, 36, 450}, + { 404, 787, -73, 1000}, { 497, 1259, 387, 1231}, + { 17, 207, 195, -79}, { 562, 358, 53, -158}, + { 493, 387, 478, 189}, { 678, 831, 640, 558}, + { -197, 523, 613, 57}, { 429, 894, 769, 111}, + { 67, 1174, 568, 511}, { 1242, 824, 251, 840}, + { 1419, 1074, 864, 481}, { 924, 1474, 669, 724}, + { 1539, 1879, 654, 1590}, { 445, 337, 1111, 541}, + { 472, 1421, 1264, 1094}, { 794, 735, 1103, 668}, + { 1055, 863, 1192, 1020}, { 778, 1105, 806, 1798}, + { 1052, 1527, 1587, 2151}, { 881, 1552, 1265, 391}, + { 726, 872, 1812, 601}, { 1469, 280, 1008, 616}, + { 1403, 577, 1803, 1244}, { 1650, 1314, 1148, 1072}, + { 1297, 1669, 1911, 1026}, { 2093, 1044, 2115, 1189}, + { 1644, 1961, 2587, 1512}, { 25, -315, -9, -106}, + { 290, -339, 428, -444}, { -68, -783, 735, 772}, + { 245, -555, 468, 47}, { 334, -895, 814, 146}, + { 235, 368, -964, -959}, { -203, 315, -1566, -1217}, + { 801, 17, -276, -354}, { 894, -495, -789, -635}, + { 716, 291, -1189, -357}, { 560, -260, -733, -2}, + { 679, -508, -1429, 211}, { -51, -62, -428, 557}, + { 322, -638, -211, 614}, { -878, -1057, -84, -71}, + { -388, -1415, -167, -318}, { -754, -1574, 214, -539}, + {-1419, -2004, -92, -787}, { -47, -856, -347, -255}, + { 23, -1211, -173, 320}, { -658, -487, -893, 353}, + { -783, -1587, -584, 507}, {-1420, -859, -378, 441}, + {-2095, -1491, -137, 439}, { -321, -1450, -1288, -12}, + { -359, -2113, -553, -8}, { -831, -1918, -1561, 32}, + {-1014, -2487, -1359, -939}, { -475, -311, -169, -236}, + { -907, -426, 276, -611}, { -96, -400, 50, -710}, + { -426, -1022, -10, -985}, { -197, -258, -744, -575}, + { -611, -930, -771, -394}, { -267, -776, -612, -939}, + { -256, -1346, -802, -1122}, { -796, -1570, -825, -754}, + { 712, 876, 141, 227}, { 981, 1509, 85, 124}, + { 1462, 1228, 979, -39}, { 1734, 999, 1481, 440}, + { 2293, 1116, 769, 440}, { 2504, 1480, 1241, 356}, + { 2474, 1909, 1558, 810}, { 917, 1134, 607, -134}, + { 509, 1809, 781, -123}, { 1712, 1506, 559, -423}, + { 2037, 2317, 726, -155}, { 3031, 2676, 1203, 331}, + { 3664, 3274, 1768, 531}, { 1610, 1839, 867, 183}, + { 1774, 1972, 1538, 97}, { 1822, 2158, 1282, 659}, + { 2222, 2758, 1818, 900}, { 3251, 2124, 1723, 996}, + { 3633, 2336, 2408, 1453}, { 2923, 3517, 2567, 1318}, +}; + +/** + * Used for the coding/decoding of the pulses positions + * for the MP-MLQ codebook + */ +static const int32_t combinatorial_table[PULSE_MAX][SUBFRAME_LEN/GRID_SIZE] = { + {118755, 98280, 80730, 65780L, 53130, + 42504, 33649, 26334, 20349, 15504, + 11628, 8568, 6188, 4368, 3003, + 2002, 1287, 792, 462, 252, + 126, 56, 21, 6, 1, + 0, 0, 0, 0, 0}, + + { 23751, 20475, 17550, 14950, 12650, + 10626, 8855, 7315, 5985, 4845, + 3876, 3060, 2380, 1820, 1365, + 1001, 715, 495, 330, 210, + 126, 70, 35, 15, 5, + 1, 0, 0, 0, 0}, + + { 3654, 3276, 2925, 2600, 2300, + 2024, 1771, 1540, 1330, 1140, + 969, 816, 680, 560, 455, + 364, 286, 220, 165, 120, + 84, 56, 35, 20, 10, + 4, 1, 0, 0, 0}, + + { 406, 378, 351, 325, 300, + 276, 253, 231, 210, 190, + 171, 153, 136, 120, 105, + 91, 78, 66, 55, 45, + 36, 28, 21, 15, 10, + 6, 3, 1, 0, 0}, + + { 29, 28, 27, 26, 25, + 24, 23, 22, 21, 20, + 19, 18, 17, 16, 15, + 14, 13, 12, 11, 10, + 9, 8, 7, 6, 5, + 4, 3, 2, 1, 0}, + + { 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1}, +}; + +static const int16_t pitch_contrib[340] = { + 60, 0, 0, 2489, 60, 0, 0, 5217, + 1, 6171, 0, 3953, 0, 10364, 1, 9357, + -1, 8843, 1, 9396, 0, 5794, -1, 10816, + 2, 11606, -2, 12072, 0, 8616, 1, 12170, + 0, 14440, 0, 7787, -1, 13721, 0, 18205, + 0, 14471, 0, 15807, 1, 15275, 0, 13480, + -1, 18375, -1, 0, 1, 11194, -1, 13010, + 1, 18836, -2, 20354, 1, 16233, -1, 0, + 60, 0, 0, 12130, 0, 13385, 1, 17834, + 1, 20875, 0, 21996, 1, 0, 1, 18277, + -1, 21321, 1, 13738, -1, 19094, -1, 20387, + -1, 0, 0, 21008, 60, 0, -2, 22807, + 0, 15900, 1, 0, 0, 17989, -1, 22259, + 1, 24395, 1, 23138, 0, 23948, 1, 22997, + 2, 22604, -1, 25942, 0, 26246, 1, 25321, + 0, 26423, 0, 24061, 0, 27247, 60, 0, + -1, 25572, 1, 23918, 1, 25930, 2, 26408, + -1, 19049, 1, 27357, -1, 24538, 60, 0, + -1, 25093, 0, 28549, 1, 0, 0, 22793, + -1, 25659, 0, 29377, 0, 30276, 0, 26198, + 1, 22521, -1, 28919, 0, 27384, 1, 30162, + -1, 0, 0, 24237, -1, 30062, 0, 21763, + 1, 30917, 60, 0, 0, 31284, 0, 29433, + 1, 26821, 1, 28655, 0, 31327, 2, 30799, + 1, 31389, 0, 32322, 1, 31760, -2, 31830, + 0, 26936, -1, 31180, 1, 30875, 0, 27873, + -1, 30429, 1, 31050, 0, 0, 0, 31912, + 1, 31611, 0, 31565, 0, 25557, 0, 31357, + 60, 0, 1, 29536, 1, 28985, -1, 26984, + -1, 31587, 2, 30836, -2, 31133, 0, 30243, + -1, 30742, -1, 32090, 60, 0, 2, 30902, + 60, 0, 0, 30027, 0, 29042, 60, 0, + 0, 31756, 0, 24553, 0, 25636, -2, 30501, + 60, 0, -1, 29617, 0, 30649, 60, 0, + 0, 29274, 2, 30415, 0, 27480, 0, 31213, + -1, 28147, 0, 30600, 1, 31652, 2, 29068, + 60, 0, 1, 28571, 1, 28730, 1, 31422, + 0, 28257, 0, 24797, 60, 0, 0, 0, + 60, 0, 0, 22105, 0, 27852, 60, 0, + 60, 0, -1, 24214, 0, 24642, 0, 23305, + 60, 0, 60, 0, 1, 22883, 0, 21601, + 60, 0, 2, 25650, 60, 0, -2, 31253, + -2, 25144, 0, 17998 +}; + +/** + * Number of non-zero pulses in the MP-MLQ excitation + */ +static const int8_t pulses[4] = {6, 5, 6, 5}; + +/** + * Size of the MP-MLQ fixed excitation codebooks + */ +static const int32_t max_pos[4] = {593775, 142506, 593775, 142506}; + +static const int16_t fixed_cb_gain[GAIN_LEVELS] = { + 1, 2, 3, 4, 6, 9, 13, 18, + 26, 38, 55, 80, 115, 166, 240, 348, + 502, 726, 1050, 1517, 2193, 3170, 4582, 6623, +}; + +static const int16_t adaptive_cb_gain85[85 * 20] = { + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 800, 1496, 167, -256, + -338, -39, -136, -1, -4, -6, -73, -8, + -15, 12, 23, 2, 16, 30, 3, -5, + -462, -686, 493, 2575, 311, -13, -28, -14, + -404, -5, -19, 13, 20, 72, 107, -77, + 8, 13, -9, -48, 1483, 144, 784, 928, + 1243, -134, -1, -37, -52, -94, -13, -71, + -6, -84, -8, -44, -112, -10, -59, -70, + -77, 275, 3522, 1056, -1254, 0, -4, -757, + -68, -95, 1, 16, -59, 4, -17, -227, + -5, 21, 269, 80, -125, -40, -264, 381, + 5027, 0, 0, -4, -8, -1542, 0, -2, + 0, 2, 0, 6, 38, 12, 81, -117, + 138, 332, 2215, 2574, 1339, -1, -6, -299, + -404, -109, -2, -18, -44, -21, -52, -348, + -11, -27, -181, -210, 3685, 2883, -887, 866, + -1639, -828, -507, -48, -45, -164, -648, 199, + 156, -194, -152, 46, 368, 288, -88, 86, + 1396, 2146, 2235, 345, 942, -118, -281, -305, + -7, -54, -182, -190, -292, -29, -45, -47, + -80, -123, -128, -19, 13, 4475, 3549, -804, + -655, 0, -1222, -768, -39, -26, -3, -2, + -969, 0, 219, 174, 0, 179, 141, -32, + -724, 254, 242, 6049, 2462, -32, -3, -3, + -2233, -370, 11, 10, -3, 267, -94, -89, + 108, -38, -36, -909, 626, -1713, 6121, 4561, + -1061, -23, -179, -2287, -1270, -68, 65, -233, + 640, -174, 477, -1704, 40, -111, 396, 295, + -350, 1391, 7985, 511, -405, -7, -118, -3892, + -15, -10, 29, 170, -678, 10, -43, -249, + -8, 34, 197, 12, 3144, -529, 608, 2530, + 3878, -603, -17, -22, -390, -918, 101, -116, + 19, -485, 81, -93, -744, 125, -144, -599, + 2589, -689, 3045, 5603, -404, -409, -29, -566, + -1916, -10, 108, -481, 128, -885, 235, -1041, + 63, -17, 75, 138, 3107, 513, 1374, -3594, + -4922, -589, -16, -115, -788, -1478, -97, -260, + -43, 681, 112, 301, 933, 154, 413, -1079, + 2468, 6010, 1107, -390, 1961, -372, -2204, -74, + -9, -234, -905, -166, -406, 58, 143, 26, + -295, -719, -132, 46, 4773, 2766, 2368, 4862, + -4044, -1390, -467, -342, -1443, -998, -806, -690, + -399, -1416, -821, -702, 1178, 682, 584, 1200, + 1665, -1879, 1443, 1701, 8562, -169, -215, -127, + -176, -4475, 190, -146, 165, -172, 195, -149, + -870, 982, -754, -889, 2716, 9011, -1007, 755, + -1785, -450, -4956, -61, -34, -194, -1493, 167, + 554, -125, -415, 46, 296, 982, -109, 82, + -2727, 7548, 1285, 938, 3420, -453, -3478, -100, + -53, -714, 1256, 213, -592, 156, -432, -73, + 569, -1576, -268, -196, 3677, 882, 4050, 1202, + 2323, -825, -47, -1001, -88, -329, -198, -909, + -218, -269, -64, -297, -521, -125, -574, -170, + 2046, -753, 122, 10102, 603, -255, -34, 0, + -6229, -22, 94, -15, 5, -1261, 464, -75, + -75, 27, -4, -372, 449, -1815, 10690, 3870, + -527, -12, -201, -6976, -914, -16, 49, -293, + 1184, -106, 428, -2525, 14, -58, 344, 124, + -941, 2352, 5049, 3650, 2637, -54, -337, -1556, + -813, -424, 135, 290, -725, 209, -524, -1125, + 151, -378, -812, -587, -1879, 796, 3117, 9569, + -404, -215, -38, -593, -5589, -9, 91, 357, + -151, 1097, -464, -1821, -46, 19, 76, 236, + -1715, 2043, -2096, 9946, 4001, -179, -254, -268, + -6038, -977, 213, -219, 261, 1041, -1240, 1272, + 418, -498, 511, -2429, -5772, -618, -3921, 284, + -3155, -2033, -23, -938, -4, -607, -218, -1381, + -148, 100, 10, 68, -1111, -119, -755, 54, + 382, 4748, 8003, -2064, 2198, -8, -1376, -3909, + -260, -294, -110, -186, -2319, 48, 598, 1008, + -51, -637, -1073, 277, -867, 3015, 11926, -1675, + 947, -45, -555, -8681, -171, -54, 159, 631, + -2195, -88, 308, 1219, 50, -174, -690, 96, + -4933, -432, 6757, 3771, 1352, -1485, -11, -2786, + -867, -111, -130, 2034, 178, 1135, 99, -1555, + 407, 35, -557, -311, 152, 9726, 4231, -1928, + 1490, -1, -5774, -1092, -226, -135, -90, -39, + -2511, 17, 1144, 498, -13, -884, -384, 175, + 2512, 193, 9033, 5361, -3148, -385, -2, -4980, + -1754, -605, -29, -1385, -106, -822, -63, -2956, + 482, 37, 1735, 1030, 8464, 2844, 12, 549, + 2132, -4373, -493, 0, -18, -277, -1469, -6, + -2, -284, -95, 0, -1101, -370, -1, -71, + 2141, -2602, 7166, 9046, -1350, -279, -413, -3134, + -4994, -111, 340, -936, 1138, -1182, 1436, -3957, + 176, -214, 590, 745, -244, 278, 13307, 1227, + -161, -3, -4, -10808, -91, -1, 4, 198, + -226, 18, -20, -997, -2, 2, 131, 12, + -1947, 8217, 6269, 917, -2559, -231, -4121, -2399, + -51, -399, 976, 745, -3144, 108, -460, -350, + -304, 1283, 979, 143, -1810, 2061, -2781, 6056, + 10058, -200, -259, -472, -2238, -6174, 227, -307, + 349, 669, -761, 1028, 1111, -1265, 1707, -3717, + 7827, 9161, -3409, 2473, -1510, -3739, -5122, -709, + -373, -139, -4376, 1628, 1906, -1181, -1382, 514, + 721, 844, -314, 228, -1430, 8313, 9541, -2955, + 1626, -124, -4218, -5556, -533, -161, 725, 832, + -4841, -257, 1499, 1721, 142, -825, -947, 293, + 2819, -4247, 5391, 8673, 2756, -485, -1101, -1774, + -4591, -463, 730, -927, 1397, -1492, 2248, -2854, + -474, 714, -907, -1459, 141, 14552, 690, 257, + -112, -1, -12926, -29, -4, 0, -125, -5, + -613, -2, -228, -10, 0, 99, 4, 1, + 11938, -1859, 1806, -962, -884, -8699, -211, -199, + -56, -47, 1355, -1316, 205, 701, -109, 106, + 644, -100, 97, -51, 3728, 1982, 2264, 4584, + 3131, -848, -239, -312, -1282, -598, -451, -515, + -273, -1043, -554, -633, -712, -378, -432, -876, + -1181, 766, 720, 14303, -216, -85, -35, -31, + -12486, -2, 55, 51, -33, 1031, -668, -628, + -15, 10, 9, 189, -4385, 4826, 10112, 1569, + 3388, -1173, -1421, -6242, -150, -700, 1291, 2706, + -2979, 420, -462, -969, 906, -998, -2091, -324, + -448, 1932, 15591, -1842, 657, -12, -227, -14837, + -207, -26, 52, 427, -1838, -50, 217, 1753, + 18, -77, -626, 74, -4141, 1844, 3962, 5517, + 6220, -1046, -207, -958, -1858, -2361, 466, 1001, + -446, 1394, -621, -1334, 1572, -700, -1504, -2094, + 729, -2299, 14755, 3657, -952, -32, -322, -13288, + -816, -55, 102, -656, 2071, -162, 513, -3294, + 42, -133, 857, 212, -1385, 5801, 13339, -3137, + 1344, -117, -2054, -10861, -600, -110, 490, 1127, + -4723, -265, 1111, 2554, 113, -476, -1094, 257, + 4710, 9661, 1073, -2467, 3274, -1354, -5697, -70, + -371, -654, -2777, -308, -633, 709, 1455, 161, + -941, -1930, -214, 493, 1843, -3624, 12422, 6898, + -1559, -207, -802, -9419, -2904, -148, 407, -1397, + 2748, -775, 1526, -5230, 175, -344, 1182, 656, + 1433, 2394, 2507, 1380, 8780, -125, -349, -383, + -116, -4705, -209, -219, -366, -120, -201, -211, + -768, -1283, -1343, -740, -1712, 12915, 5883, -2197, + 991, -179, -10181, -2112, -294, -60, 1350, 615, + -4638, -229, 1732, 789, 103, -781, -356, 133, + 15072, 2158, -1245, 910, -496, -13865, -284, -94, + -50, -15, -1986, 1145, 164, -837, -119, 69, + 456, 65, -37, 27, 4655, 7319, 4916, 586, + -3381, -1322, -3270, -1475, -20, -697, -2079, -1396, + -2196, -166, -261, -175, 960, 1510, 1014, 120, + 1191, -2140, 5120, 13498, -1418, -86, -279, -1600, + -11121, -122, 155, -372, 669, -981, 1763, -4218, + 103, -185, 443, 1168, -1530, -817, 8191, 9632, + -1452, -143, -40, -4095, -5663, -128, -76, 765, + 408, 900, 480, -4815, -135, -72, 726, 854, + -3236, 607, 1696, -2106, 11485, -639, -22, -175, + -270, -8051, 119, 335, -62, -416, 78, 218, + 2268, -425, -1189, 1476, 3203, -1903, -837, 9679, + 7057, -626, -221, -42, -5718, -3039, 372, 163, + -97, -1892, 1124, 494, -1380, 819, 360, -4169, + 213, -655, 17015, 620, -384, -2, -26, -17671, + -23, -9, 8, -221, 681, -8, 24, -644, + 5, -15, 399, 14, 5088, 35, -3339, 3726, + 8488, -1580, 0, -680, -847, -4397, -10, 1037, + 7, -1157, -8, 759, -2636, -18, 1730, -1930, + -988, 1454, -2688, 15039, 2682, -59, -129, -441, + -13805, -439, 87, -162, 238, 907, -1335, 2467, + 161, -238, 440, -2462, -4865, -2842, -53, 5495, + 6523, -1445, -493, 0, -1843, -2597, -844, -16, + -9, 1632, 953, 18, 1937, 1131, 21, -2188, + 3076, 15069, -2914, 1810, -971, -577, -13860, -518, + -200, -57, -2829, 547, 2680, -339, -1665, 322, + 182, 893, -172, 107, 1311, 5355, 11054, 2299, + -3654, -105, -1750, -7458, -322, -814, -428, -885, + -3613, -184, -751, -1551, 292, 1194, 2465, 512, + 4035, 5619, 4618, 1815, 1912, -994, -1927, -1301, + -201, -223, -1384, -1137, -1583, -447, -622, -511, + -471, -656, -539, -211, -2131, 2754, -4501, 12879, + 7432, -277, -463, -1236, -10124, -3371, 358, -585, + 756, 1675, -2165, 3538, 967, -1249, 2042, -5842, + 5618, -515, 3219, -4149, 4857, -1926, -16, -632, + -1050, -1440, 176, -1104, 101, 1422, -130, 815, + -1666, 152, -954, 1230, 1838, -1709, 1139, 16867, + 716, -206, -178, -79, -17366, -31, 191, -127, + 118, -1892, 1759, -1173, -80, 74, -49, -737, + 1978, -3845, 10050, 11854, -2492, -238, -902, -6164, + -8576, -379, 464, -1213, 2358, -1431, 2782, -7271, + 301, -585, 1529, 1803, -2600, 11246, 11289, -3647, + 1463, -412, -7720, -7778, -812, -130, 1784, 1791, + -7749, -578, 2504, 2513, 232, -1004, -1008, 325, + 3442, 907, 2725, 8970, 3638, -723, -50, -453, + -4911, -808, -190, -572, -150, -1884, -496, -1492, + -764, -201, -605, -1992, -126, 17498, 3481, -2003, + 1090, 0, -18689, -739, -244, -72, 135, 26, + -3717, -15, 2139, 425, 8, -1165, -231, 133, + -1814, 1048, -2164, 4070, 16272, -200, -67, -285, + -1011, -16160, 116, -239, 138, 450, -260, 537, + 1801, -1041, 2149, -4042, 9354, 12580, -1883, 962, + -617, -5341, -9660, -216, -56, -23, -7183, 1075, + 1446, -549, -738, 110, 352, 474, -71, 36, + 1708, 4199, 7387, 6335, 1003, -178, -1076, -3330, + -2449, -61, -437, -770, -1893, -660, -1623, -2856, + -104, -257, -452, -388, -2624, 5623, 17310, -2353, + 592, -420, -1930, -18288, -338, -21, 900, 2772, + -5941, -376, 807, 2486, 94, -203, -625, 85, + 1211, -850, 1193, -1926, 15992, -89, -44, -86, + -226, -15609, 62, -88, 61, 142, -100, 140, + -1182, 830, -1165, 1880, 3983, -2054, 11506, -19, + 3622, -968, -257, -8080, 0, -801, 499, -2797, + 1442, 4, -2, 13, -880, 454, -2544, 4, + -786, -1354, 16092, 7246, -1665, -37, -111, -15805, + -3205, -169, -65, 772, 1330, 348, 599, -7117, + -80, -137, 1636, 736, -4316, -511, 6674, 11665, + 4633, -1137, -15, -2719, -8305, -1310, -134, 1758, + 208, 3073, 364, -4752, 1220, 144, -1887, -3299, + 7912, 4557, 1937, 1885, 7037, -3821, -1267, -229, + -216, -3022, -2200, -935, -538, -910, -524, -222, + -3398, -1957, -832, -809, 3434, 2967, 5867, 8196, + 8766, -720, -537, -2101, -4100, -4690, -622, -1230, + -1062, -1718, -1484, -2935, -1837, -1588, -3139, -4385, + 5881, 9176, 8119, 3934, 3355, -2111, -5139, -4023, + -944, -687, -3294, -2914, -4547, -1412, -2203, -1949, + -1204, -1879, -1662, -805 +}; + +static const int16_t adaptive_cb_gain170[170 * 20] = { + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 776, 212, 715, 670, + 809, -36, -2, -31, -27, -39, -10, -33, + -9, -31, -8, -29, -38, -10, -35, -33, + 1296, 1316, -168, -320, -815, -102, -105, -1, + -6, -40, -104, 13, 13, 25, 25, -3, + 64, 65, -8, -15, -589, 680, 2478, 308, + -596, -21, -28, -375, -5, -21, 24, 89, + -102, 11, -12, -46, -21, 24, 90, 11, + -735, -487, -5, 2948, 468, -33, -14, 0, + -530, -13, -21, 0, 0, 132, 87, 0, + 21, 13, 0, -84, 1042, 1730, 1068, 333, + 626, -66, -182, -69, -6, -23, -110, -67, + -112, -21, -35, -21, -39, -66, -40, -12, + 486, -769, 4074, 2825, -1107, -14, -36, -1013, + -487, -74, 22, -120, 191, -83, 132, -702, + 32, -52, 275, 191, 1521, -767, -124, 4320, + 1026, -141, -35, 0, -1139, -64, 71, 11, + -5, -401, 202, 32, -95, 48, 7, -270, + 2425, 1267, 3439, -91, -1166, -359, -98, -722, + 0, -83, -187, -509, -266, 13, 7, 19, + 172, 90, 244, -6, -1251, 975, 173, 4039, + 2005, -95, -58, -1, -996, -245, 74, 13, + -10, 308, -240, -42, 153, -119, -21, -494, + 1820, 632, 1322, 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11535, 2203, -452, 7169, -1954, -8121, -296, + -12, -3137, -3984, -761, -1551, 156, 318, 60, + -2476, -5048, -964, 197, 2914, -2914, 3485, -3965, + 13675, -518, -518, -741, -959, -11414, 518, -620, + 620, 705, -705, 843, -2433, 2432, -2909, 3310, + 7843, 1907, 1022, 8882, 7972, -3755, -222, -63, + -4815, -3879, -913, -489, -119, -4252, -1034, -554, + -3816, -928, -497, -4322, 13807, 9531, 1436, 1612, + 1779, -11636, -5544, -125, -158, -193, -8032, -1210, + -835, -1358, -938, -141, -1499, -1035, -156, -175, + 13620, -5337, 5450, -2263, 1723, -11322, -1738, -1813, + -312, -181, 4436, -4531, 1775, 1881, -737, 752, + -1432, 561, -573, 238, 5297, 8374, 8872, 7694, + 6538, -1712, -4280, -4804, -3613, -2609, -2707, -2868, + -4534, -2487, -3932, -4166, -2113, -3341, -3540, -3070 +}; + +/** + * 0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15 + */ +static const int16_t postfilter_tbl[2][LPC_ORDER] = { + /* Zero */ + {21299, 13844, 8999, 5849, 3802, 2471, 1606, 1044, 679, 441}, + /* Pole */ + {24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845} +};

3 2

swresample: check for invalid sample formats.
by git＠videolan.org 30 Sep '11

30 Sep '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Sat Oct 1 01:39:17 2011 +0200| [834b3760a7ca112573e813bd6c3573a8c0daf4ed] | committer: Michael Niedermayer swresample: check for invalid sample formats. Bug-Found-by: Justin Ruggles Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=834b3760a7ca11257… --- libswresample/swresample.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/libswresample/swresample.c b/libswresample/swresample.c index 519cadf..7276c7a 100644 --- a/libswresample/swresample.c +++ b/libswresample/swresample.c @@ -144,7 +144,15 @@ int swr_init(SwrContext *s){ s-> in_sample_fmt &= 0xFF; s->out_sample_fmt &= 0xFF; - //We assume AVOptions checked the various values and the defaults where allowed + 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));

1 0

avformat_free_context: favor av_freep()
by git＠videolan.org 30 Sep '11

30 Sep '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Sat Oct 1 01:29:30 2011 +0200| [2a93f28a4b6eef8b93046e0b4e3225f2ff1e7324] | committer: Michael Niedermayer avformat_free_context: favor av_freep() Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2a93f28a4b6eef8b9… --- libavformat/utils.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/libavformat/utils.c b/libavformat/utils.c index 0d88991..93a3f69 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -2713,13 +2713,13 @@ void avformat_free_context(AVFormatContext *s) av_free_packet(&st->cur_pkt); } av_dict_free(&st->metadata); - av_free(st->index_entries); - av_free(st->codec->extradata); - av_free(st->codec->subtitle_header); - av_free(st->codec); - av_free(st->priv_data); - av_free(st->info); - av_free(st); + av_freep(&st->index_entries); + av_freep(&st->codec->extradata); + av_freep(&st->codec->subtitle_header); + av_freep(&st->codec); + av_freep(&st->priv_data); + av_freep(&st->info); + av_freep(&st); } for(i=s->nb_programs-1; i>=0; i--) { av_dict_free(&s->programs[i]->metadata); @@ -2730,7 +2730,7 @@ void avformat_free_context(AVFormatContext *s) av_freep(&s->priv_data); while(s->nb_chapters--) { av_dict_free(&s->chapters[s->nb_chapters]->metadata); - av_free(s->chapters[s->nb_chapters]); + av_freep(&s->chapters[s->nb_chapters]); } av_freep(&s->chapters); av_dict_free(&s->metadata);

1 0

oggenc: favor av_freep()
by git＠videolan.org 30 Sep '11

30 Sep '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Sat Oct 1 01:26:55 2011 +0200| [3b6c0ba382cc261b81f1451366e56a7878d4919a] | committer: Michael Niedermayer oggenc: favor av_freep() Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3b6c0ba382cc261b8… --- libavformat/oggenc.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/libavformat/oggenc.c b/libavformat/oggenc.c index 21b3501..a527ab3 100644 --- a/libavformat/oggenc.c +++ b/libavformat/oggenc.c @@ -516,11 +516,11 @@ static int ogg_write_trailer(AVFormatContext *s) OGGStreamContext *oggstream = st->priv_data; if (st->codec->codec_id == CODEC_ID_FLAC || st->codec->codec_id == CODEC_ID_SPEEX) { - av_free(oggstream->header[0]); - av_free(oggstream->header[1]); + av_freep(&oggstream->header[0]); + av_freep(&oggstream->header[1]); } else - av_free(oggstream->header[1]); + av_freep(&oggstream->header[1]); av_freep(&st->priv_data); } return 0;

1 0

swr_test: test planar audio
by git＠videolan.org 30 Sep '11

30 Sep '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Thu Sep 29 04:55:53 2011 +0200| [17d51938d23c5f60f1ae0f4944724944c1b9dfdd] | committer: Michael Niedermayer swr_test: test planar audio Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=17d51938d23c5f60f… --- libswresample/swresample_test.c | 61 +++++++++++++++++++++++++++++++------- 1 files changed, 49 insertions(+), 12 deletions(-) diff --git a/libswresample/swresample_test.c b/libswresample/swresample_test.c index e8683cd..61e1b09 100644 --- a/libswresample/swresample_test.c +++ b/libswresample/swresample_test.c @@ -28,7 +28,16 @@ #define ASSERT_LEVEL 2 -static double get(const void *p, int index, enum AVSampleFormat f){ +static double get(const uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f){ + const uint8_t *p; + if(f>=0x100){ + f&=0xFF; + p= a[ch]; + }else{ + p= a[0]; + index= ch + index*ch_count; + } + switch(f){ case AV_SAMPLE_FMT_U8 : return ((const uint8_t*)p)[index]/255.0*2-1.0; case AV_SAMPLE_FMT_S16: return ((const int16_t*)p)[index]/32767.0; @@ -39,7 +48,15 @@ static double get(const void *p, int index, enum AVSampleFormat f){ } } -static void set(void *p, int index, enum AVSampleFormat f, double v){ +static void set(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f, double v){ + uint8_t *p; + if(f>=0x100){ + f&=0xFF; + p= a[ch]; + }else{ + p= a[0]; + index= ch + index*ch_count; + } switch(f){ case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= (v+1.0)*255.0/2; break; case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= v*32767; break; @@ -68,6 +85,19 @@ AV_CH_LAYOUT_7POINT1_WIDE , 0 }; +static void setup_array(uint8_t *out[SWR_CH_MAX], uint8_t *in, enum AVSampleFormat format, int samples){ + if(format >= 0x100){ + int i; + int plane_size= av_get_bytes_per_sample(format&0xFF)*samples; + format&=0xFF; + for(i=0; i<SWR_CH_MAX; i++){ + out[i]= in + i*plane_size; + } + }else{ + out[0]= in; + } +} + int main(int argc, char **argv){ int in_sample_rate, out_sample_rate, ch ,i, in_ch_layout_index, out_ch_layout_index, osr; uint64_t in_ch_layout, out_ch_layout; @@ -76,6 +106,11 @@ int main(int argc, char **argv){ uint8_t array_in[SAMPLES*8*8]; uint8_t array_mid[SAMPLES*8*8*3]; uint8_t array_out[SAMPLES*8*8+100]; + uint8_t *ain[SWR_CH_MAX]; + uint8_t *aout[SWR_CH_MAX]; + uint8_t *amid[SWR_CH_MAX]; + int planar_in=256, planar_out=256; + struct SwrContext * forw_ctx= NULL; struct SwrContext *backw_ctx= NULL; @@ -95,10 +130,10 @@ int main(int argc, char **argv){ in_ch_count, out_ch_count, in_sample_rate, out_sample_rate, av_get_sample_fmt_name(in_sample_fmt), av_get_sample_fmt_name(out_sample_fmt)); - forw_ctx = swr_alloc2(forw_ctx, out_ch_layout, out_sample_fmt, out_sample_rate, - in_ch_layout, in_sample_fmt, in_sample_rate, 0, 0); + forw_ctx = swr_alloc2(forw_ctx, out_ch_layout, out_sample_fmt+planar_out, out_sample_rate, + in_ch_layout, in_sample_fmt+planar_in , in_sample_rate, 0, 0); backw_ctx = swr_alloc2(backw_ctx,in_ch_layout, in_sample_fmt, in_sample_rate, - out_ch_layout, out_sample_fmt, out_sample_rate, 0, 0); + out_ch_layout, out_sample_fmt+planar_out, out_sample_rate, 0, 0); if(swr_init( forw_ctx) < 0) fprintf(stderr, "swr_init(->) failed\n"); if(swr_init(backw_ctx) < 0) @@ -108,14 +143,16 @@ int main(int argc, char **argv){ if(!backw_ctx) fprintf(stderr, "Failed to init backw_ctx\n"); //FIXME test planar + setup_array(ain , array_in , in_sample_fmt+planar_in , SAMPLES); + setup_array(amid, array_mid, out_sample_fmt+planar_out, 3*SAMPLES); + setup_array(aout, array_out, in_sample_fmt , SAMPLES); for(ch=0; ch<in_ch_count; ch++){ for(i=0; i<SAMPLES; i++) - set(array_in, ch + i*in_ch_count, in_sample_fmt, sin(i*i*3/SAMPLES)); + set(ain, ch, i, in_ch_count, in_sample_fmt+planar_in, sin(i*i*3/SAMPLES)); } - mid_count= swr_convert(forw_ctx, ( uint8_t*[]){array_mid}, 3*SAMPLES, - (const uint8_t*[]){array_in }, SAMPLES); - out_count= swr_convert(backw_ctx,( uint8_t*[]){array_out}, 3*SAMPLES, - (const uint8_t*[]){array_mid}, mid_count); + mid_count= swr_convert(forw_ctx, amid, 3*SAMPLES, ain, SAMPLES); + out_count= swr_convert(backw_ctx,aout, SAMPLES, amid, mid_count); + for(ch=0; ch<in_ch_count; ch++){ double sse, x, maxdiff=0; double sum_a= 0; @@ -124,8 +161,8 @@ int main(int argc, char **argv){ double sum_bb= 0; double sum_ab= 0; for(i=0; i<out_count; i++){ - double a= get(array_in , ch + i*in_ch_count, in_sample_fmt); - double b= get(array_out, ch + i*in_ch_count, in_sample_fmt); + double a= get(ain , ch, i, in_ch_count, in_sample_fmt+planar_in); + double b= get(aout, ch, i, in_ch_count, in_sample_fmt); sum_a += a; sum_b += b; sum_aa+= a*a;

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swr: minor fixes to get planar audio working
by git＠videolan.org 30 Sep '11

30 Sep '11

ffmpeg | branch: master | Michael Niedermayer <michaelni(a)gmx.at> | Thu Sep 29 04:53:50 2011 +0200| [a75b254c74d71414f15764979e718e74cd90f5aa] | committer: Michael Niedermayer swr: minor fixes to get planar audio working Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a75b254c74d71414f… --- libswresample/swresample.c | 32 ++++++++++++++++++++++---------- 1 files changed, 22 insertions(+), 10 deletions(-) diff --git a/libswresample/swresample.c b/libswresample/swresample.c index 20a1e64..519cadf 100644 --- a/libswresample/swresample.c +++ b/libswresample/swresample.c @@ -39,10 +39,10 @@ static const AVOption options[]={ {"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, 0}, -{"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 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"}, @@ -139,6 +139,11 @@ int swr_init(SwrContext *s){ swr_audio_convert_free(&s-> in_convert); swr_audio_convert_free(&s->out_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; + //We assume AVOptions checked the various values and the defaults where allowed if( s->int_sample_fmt != AV_SAMPLE_FMT_S16 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){ @@ -250,6 +255,17 @@ static void copy(AudioData *out, AudioData *in, 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; @@ -264,12 +280,8 @@ int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_coun out_count = in_count; } - av_assert0(in ->planar == 0); - av_assert0(out->planar == 0); - for(i=0; i<s-> in.ch_count; i++) - in ->ch[i]= in_arg[0] + i* in->bps; - for(i=0; i<s->out.ch_count; i++) - out->ch[i]= out_arg[0] + i*out->bps; + fill_audiodata(in , in_arg); + fill_audiodata(out, out_arg); // 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);

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