diff --git a/configure b/configure index b4e433cd20..aa45fd77ff 100755 --- a/configure +++ b/configure @@ -1648,6 +1648,7 @@ tls_protocol_select="tcp_protocol" udp_protocol_deps="network" # filters +aconvert_filter_deps="swresample" amovie_filter_deps="avcodec avformat" aresample_filter_deps="swresample" ass_filter_deps="libass" diff --git a/doc/filters.texi b/doc/filters.texi index 1f3522e08d..c64f899011 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -104,17 +104,15 @@ Below is a description of the currently available audio filters. Convert the input audio format to the specified formats. The filter accepts a string of the form: -"@var{sample_format}:@var{channel_layout}:@var{packing_format}". +"@var{sample_format}:@var{channel_layout}". -@var{sample_format} specifies the sample format, and can be a string or -the corresponding numeric value defined in @file{libavutil/samplefmt.h}. +@var{sample_format} specifies the sample format, and can be a string or the +corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p' +suffix for a planar sample format. @var{channel_layout} specifies the channel layout, and can be a string or the corresponding number value defined in @file{libavutil/audioconvert.h}. -@var{packing_format} specifies the type of packing in output, can be one -of "planar" or "packed", or the corresponding numeric values "0" or "1". - The special parameter "auto", signifies that the filter will automatically select the output format depending on the output filter. @@ -122,16 +120,15 @@ Some examples follow. @itemize @item -Convert input to unsigned 8-bit, stereo, packed: +Convert input to float, planar, stereo: @example -aconvert=u8:stereo:packed +aconvert=fltp:stereo @end example @item -Convert input to unsigned 8-bit, automatically select out channel layout -and packing format: +Convert input to unsigned 8-bit, automatically select out channel layout: @example -aconvert=u8:auto:auto +aconvert=u8:auto @end example @end itemize diff --git a/libavfilter/Makefile b/libavfilter/Makefile index ff0ba755f9..9fbb59b639 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -3,7 +3,7 @@ include $(SUBDIR)../config.mak NAME = avfilter FFLIBS = avutil -FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec +FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec diff --git a/libavfilter/af_aconvert.c b/libavfilter/af_aconvert.c index e3c7f8cc3d..8c1b5dc346 100644 --- a/libavfilter/af_aconvert.c +++ b/libavfilter/af_aconvert.c @@ -23,98 +23,19 @@ /** * @file * sample format and channel layout conversion audio filter - * based on code in libavcodec/resample.c by Fabrice Bellard and - * libavcodec/audioconvert.c by Michael Niedermayer */ -#include "libavutil/audioconvert.h" #include "libavutil/avstring.h" -#include "libavcodec/audioconvert.h" +#include "libswresample/swresample.h" #include "avfilter.h" #include "internal.h" typedef struct { - enum AVSampleFormat out_sample_fmt, in_sample_fmt; ///< in/out sample formats - int64_t out_chlayout, in_chlayout; ///< in/out channel layout - int out_nb_channels, in_nb_channels; ///< number of in/output channels - enum AVFilterPacking out_packing_fmt, in_packing_fmt; ///< output packing format - - int max_nb_samples; ///< maximum number of buffered samples - AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer - AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions - - uint8_t *in_mix[8], *out_mix[8]; ///< input/output for rematrixing functions - uint8_t *packed_data[8]; ///< pointers for packing conversion - int out_strides[8], in_strides[8]; ///< input/output strides for av_audio_convert - uint8_t **in_conv, **out_conv; ///< input/output for av_audio_convert - - AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format - - void (*convert_chlayout)(); ///< function to do the requested rematrixing + enum AVSampleFormat out_sample_fmt; + int64_t out_chlayout; + struct SwrContext *swr; } AConvertContext; -#define REMATRIX_FUNC_SIG(NAME) static void REMATRIX_FUNC_NAME(NAME) \ - (FMT_TYPE *outp[], FMT_TYPE *inp[], int nb_samples, AConvertContext *aconvert) - -#define FMT_TYPE uint8_t -#define REMATRIX_FUNC_NAME(NAME) NAME ## _u8 -#include "af_aconvert_rematrix.c" - -#define FMT_TYPE int16_t -#define REMATRIX_FUNC_NAME(NAME) NAME ## _s16 -#include "af_aconvert_rematrix.c" - -#define FMT_TYPE int32_t -#define REMATRIX_FUNC_NAME(NAME) NAME ## _s32 -#include "af_aconvert_rematrix.c" - -#define FLOATING - -#define FMT_TYPE float -#define REMATRIX_FUNC_NAME(NAME) NAME ## _flt -#include "af_aconvert_rematrix.c" - -#define FMT_TYPE double -#define REMATRIX_FUNC_NAME(NAME) NAME ## _dbl -#include "af_aconvert_rematrix.c" - -#define FMT_TYPE uint8_t -#define REMATRIX_FUNC_NAME(NAME) NAME -REMATRIX_FUNC_SIG(stereo_remix_planar) -{ - int size = av_get_bytes_per_sample(aconvert->in_sample_fmt) * nb_samples; - - memcpy(outp[0], inp[0], size); - memcpy(outp[1], inp[aconvert->in_nb_channels == 1 ? 0 : 1], size); -} - -#define REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC, PACKING) \ - {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_U8, FUNC##_u8}, \ - {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S16, FUNC##_s16}, \ - {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S32, FUNC##_s32}, \ - {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_FLT, FUNC##_flt}, \ - {INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_DBL, FUNC##_dbl}, - -#define REGISTER_FUNC(INCHLAYOUT, OUTCHLAYOUT, FUNC) \ - REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_packed, AVFILTER_PACKED) \ - REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_planar, AVFILTER_PLANAR) - -static const struct RematrixFunctionInfo { - int64_t in_chlayout, out_chlayout; - int planar, sfmt; - void (*func)(); -} rematrix_funcs[] = { - REGISTER_FUNC (AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, stereo_to_surround_5p1) - REGISTER_FUNC (AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_5p1_to_stereo) - REGISTER_FUNC_PACKING(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, stereo_to_mono_packed, AVFILTER_PACKED) - REGISTER_FUNC_PACKING(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, mono_to_stereo_packed, AVFILTER_PACKED) - REGISTER_FUNC (0, AV_CH_LAYOUT_MONO, mono_downmix) - REGISTER_FUNC_PACKING(0, AV_CH_LAYOUT_STEREO, stereo_downmix_packed, AVFILTER_PACKED) - - // This function works for all sample formats - {0, AV_CH_LAYOUT_STEREO, AVFILTER_PLANAR, -1, stereo_remix_planar} -}; - static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque) { AConvertContext *aconvert = ctx->priv; @@ -124,7 +45,6 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque) aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE; aconvert->out_chlayout = 0; - aconvert->out_packing_fmt = -1; if ((arg = av_strtok(args, ":", &ptr)) && strcmp(arg, "auto")) { if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0) @@ -134,10 +54,6 @@ static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque) if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0) goto end; } - if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) { - if ((ret = ff_parse_packing_format((int *)&aconvert->out_packing_fmt, arg, ctx)) < 0) - goto end; - } end: av_freep(&args); @@ -147,10 +63,7 @@ end: static av_cold void uninit(AVFilterContext *ctx) { AConvertContext *aconvert = ctx->priv; - avfilter_unref_buffer(aconvert->mix_samplesref); - avfilter_unref_buffer(aconvert->out_samplesref); - if (aconvert->audioconvert_ctx) - av_audio_convert_free(aconvert->audioconvert_ctx); + swr_free(&aconvert->swr); } static int query_formats(AVFilterContext *ctx) @@ -159,6 +72,7 @@ static int query_formats(AVFilterContext *ctx) AConvertContext *aconvert = ctx->priv; AVFilterLink *inlink = ctx->inputs[0]; AVFilterLink *outlink = ctx->outputs[0]; + int out_packing = av_sample_fmt_is_planar(aconvert->out_sample_fmt); avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO), &inlink->out_formats); @@ -182,219 +96,64 @@ static int query_formats(AVFilterContext *ctx) avfilter_formats_ref(avfilter_make_all_packing_formats(), &inlink->out_packing); - if (aconvert->out_packing_fmt != -1) { - formats = NULL; - avfilter_add_format(&formats, aconvert->out_packing_fmt); - avfilter_formats_ref(formats, &outlink->in_packing); - } else - avfilter_formats_ref(avfilter_make_all_packing_formats(), - &outlink->in_packing); + formats = NULL; + avfilter_add_format(&formats, out_packing); + avfilter_formats_ref(formats, &outlink->in_packing); return 0; } static int config_output(AVFilterLink *outlink) { - AVFilterLink *inlink = outlink->src->inputs[0]; - AConvertContext *aconvert = outlink->src->priv; + int ret; + AVFilterContext *ctx = outlink->src; + AVFilterLink *inlink = ctx->inputs[0]; + AConvertContext *aconvert = ctx->priv; char buf1[64], buf2[64]; - aconvert->in_sample_fmt = inlink->format; - aconvert->in_packing_fmt = inlink->planar; - if (aconvert->out_packing_fmt == -1) - aconvert->out_packing_fmt = outlink->planar; - aconvert->in_chlayout = inlink->channel_layout; - aconvert->in_nb_channels = - av_get_channel_layout_nb_channels(inlink->channel_layout); - /* if not specified in args, use the format and layout of the output */ if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE) aconvert->out_sample_fmt = outlink->format; if (aconvert->out_chlayout == 0) aconvert->out_chlayout = outlink->channel_layout; - aconvert->out_nb_channels = - av_get_channel_layout_nb_channels(outlink->channel_layout); + + aconvert->swr = swr_alloc_set_opts(aconvert->swr, + aconvert->out_chlayout, aconvert->out_sample_fmt, inlink->sample_rate, + inlink->channel_layout, inlink->format, inlink->sample_rate, + 0, ctx); + if (!aconvert->swr) + return AVERROR(ENOMEM); + ret = swr_init(aconvert->swr); + if (ret < 0) + return ret; av_get_channel_layout_string(buf1, sizeof(buf1), -1, inlink ->channel_layout); av_get_channel_layout_string(buf2, sizeof(buf2), -1, outlink->channel_layout); - av_log(outlink->src, AV_LOG_INFO, + av_log(ctx, AV_LOG_INFO, "fmt:%s cl:%s planar:%i -> fmt:%s cl:%s planar:%i\n", av_get_sample_fmt_name(inlink ->format), buf1, inlink ->planar, av_get_sample_fmt_name(outlink->format), buf2, outlink->planar); - /* compute which channel layout conversion to use */ - if (inlink->channel_layout != outlink->channel_layout) { - int i; - for (i = 0; i < sizeof(rematrix_funcs); i++) { - const struct RematrixFunctionInfo *f = &rematrix_funcs[i]; - if ((f->in_chlayout == 0 || f->in_chlayout == inlink ->channel_layout) && - (f->out_chlayout == 0 || f->out_chlayout == outlink->channel_layout) && - (f->planar == -1 || f->planar == inlink->planar) && - (f->sfmt == -1 || f->sfmt == inlink->format) - ) { - aconvert->convert_chlayout = f->func; - break; - } - } - if (!aconvert->convert_chlayout) { - av_log(outlink->src, AV_LOG_ERROR, - "Unsupported channel layout conversion '%s -> %s' requested!\n", - buf1, buf2); - return AVERROR(EINVAL); - } - } - return 0; } -static int init_buffers(AVFilterLink *inlink, int nb_samples) -{ - AConvertContext *aconvert = inlink->dst->priv; - AVFilterLink * const outlink = inlink->dst->outputs[0]; - int i, packed_stride = 0; - const unsigned - packing_conv = inlink->planar != outlink->planar && - aconvert->out_nb_channels != 1, - format_conv = inlink->format != outlink->format; - int nb_channels = aconvert->out_nb_channels; - - uninit(inlink->dst); - aconvert->max_nb_samples = nb_samples; - - if (aconvert->convert_chlayout) { - /* allocate buffer for storing intermediary mixing samplesref */ - uint8_t *data[8]; - int linesize[8]; - int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout); - - if (av_samples_alloc(data, linesize, nb_channels, nb_samples, - inlink->format, 16) < 0) - goto fail_no_mem; - aconvert->mix_samplesref = - avfilter_get_audio_buffer_ref_from_arrays(data, linesize, AV_PERM_WRITE, - nb_samples, inlink->format, - outlink->channel_layout, - inlink->planar); - if (!aconvert->mix_samplesref) - goto fail_no_mem; - } - - // if there's a format/packing conversion we need an audio_convert context - if (format_conv || packing_conv) { - aconvert->out_samplesref = - avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples); - if (!aconvert->out_samplesref) - goto fail_no_mem; - - aconvert->in_strides [0] = av_get_bytes_per_sample(inlink ->format); - aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format); - - aconvert->out_conv = aconvert->out_samplesref->data; - if (aconvert->mix_samplesref) - aconvert->in_conv = aconvert->mix_samplesref->data; - - if (packing_conv) { - // packed -> planar - if (outlink->planar == AVFILTER_PLANAR) { - if (aconvert->mix_samplesref) - aconvert->packed_data[0] = aconvert->mix_samplesref->data[0]; - aconvert->in_conv = aconvert->packed_data; - packed_stride = aconvert->in_strides[0]; - aconvert->in_strides[0] *= nb_channels; - // planar -> packed - } else { - aconvert->packed_data[0] = aconvert->out_samplesref->data[0]; - aconvert->out_conv = aconvert->packed_data; - packed_stride = aconvert->out_strides[0]; - aconvert->out_strides[0] *= nb_channels; - } - } else if (outlink->planar == AVFILTER_PACKED) { - /* If there's no packing conversion, and the stream is packed - * then we treat the entire stream as one big channel - */ - nb_channels = 1; - } - - for (i = 1; i < nb_channels; i++) { - aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride; - aconvert->in_strides[i] = aconvert->in_strides[0]; - aconvert->out_strides[i] = aconvert->out_strides[0]; - } - - aconvert->audioconvert_ctx = - av_audio_convert_alloc(outlink->format, nb_channels, - inlink->format, nb_channels, NULL, 0); - if (!aconvert->audioconvert_ctx) - goto fail_no_mem; - } - - return 0; - -fail_no_mem: - av_log(inlink->dst, AV_LOG_ERROR, "Could not allocate memory.\n"); - return AVERROR(ENOMEM); -} - static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) { AConvertContext *aconvert = inlink->dst->priv; - AVFilterBufferRef *curbuf = insamplesref; - AVFilterLink * const outlink = inlink->dst->outputs[0]; - int chan_mult; + const int n = insamplesref->audio->nb_samples; + AVFilterLink *const outlink = inlink->dst->outputs[0]; + AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n); - /* in/reinint the internal buffers if this is the first buffer - * provided or it is needed to use a bigger one */ - if (!aconvert->max_nb_samples || - (curbuf->audio->nb_samples > aconvert->max_nb_samples)) - if (init_buffers(inlink, curbuf->audio->nb_samples) < 0) { - av_log(inlink->dst, AV_LOG_ERROR, "Could not initialize buffers.\n"); - return; - } + swr_convert(aconvert->swr, outsamplesref->data, n, + (void *)insamplesref->data, n); - /* if channel mixing is required */ - if (aconvert->mix_samplesref) { - memcpy(aconvert->in_mix, curbuf->data, sizeof(aconvert->in_mix)); - memcpy(aconvert->out_mix, aconvert->mix_samplesref->data, sizeof(aconvert->out_mix)); - aconvert->convert_chlayout(aconvert->out_mix, - aconvert->in_mix, - curbuf->audio->nb_samples, - aconvert); - curbuf = aconvert->mix_samplesref; - } + avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); + outsamplesref->audio->channel_layout = outlink->channel_layout; + outsamplesref->audio->planar = outlink->planar; - if (aconvert->audioconvert_ctx) { - if (!aconvert->mix_samplesref) { - if (aconvert->in_conv == aconvert->packed_data) { - int i, packed_stride = av_get_bytes_per_sample(inlink->format); - aconvert->packed_data[0] = curbuf->data[0]; - for (i = 1; i < aconvert->out_nb_channels; i++) - aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride; - } else { - aconvert->in_conv = curbuf->data; - } - } - - chan_mult = inlink->planar == outlink->planar && inlink->planar == 0 ? - aconvert->out_nb_channels : 1; - - av_audio_convert(aconvert->audioconvert_ctx, - (void * const *) aconvert->out_conv, - aconvert->out_strides, - (const void * const *) aconvert->in_conv, - aconvert->in_strides, - curbuf->audio->nb_samples * chan_mult); - - curbuf = aconvert->out_samplesref; - } - - avfilter_copy_buffer_ref_props(curbuf, insamplesref); - curbuf->audio->channel_layout = outlink->channel_layout; - curbuf->audio->planar = outlink->planar; - - avfilter_filter_samples(inlink->dst->outputs[0], - avfilter_ref_buffer(curbuf, ~0)); + avfilter_filter_samples(outlink, outsamplesref); avfilter_unref_buffer(insamplesref); } diff --git a/libavfilter/af_aconvert_rematrix.c b/libavfilter/af_aconvert_rematrix.c deleted file mode 100644 index d75ca5aa40..0000000000 --- a/libavfilter/af_aconvert_rematrix.c +++ /dev/null @@ -1,172 +0,0 @@ -/* - * Copyright (c) 2011 Mina Nagy Zaki - * - * 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 rematrixing functions, based on functions from libavcodec/resample.c - */ - -#if defined(FLOATING) -# define DIV2 /2 -#else -# define DIV2 >>1 -#endif - -REMATRIX_FUNC_SIG(stereo_to_mono_packed) -{ - while (nb_samples >= 4) { - outp[0][0] = (inp[0][0] + inp[0][1]) DIV2; - outp[0][1] = (inp[0][2] + inp[0][3]) DIV2; - outp[0][2] = (inp[0][4] + inp[0][5]) DIV2; - outp[0][3] = (inp[0][6] + inp[0][7]) DIV2; - outp[0] += 4; - inp[0] += 8; - nb_samples -= 4; - } - while (nb_samples--) { - outp[0][0] = (inp[0][0] + inp[0][1]) DIV2; - outp[0]++; - inp[0] += 2; - } -} - -REMATRIX_FUNC_SIG(stereo_downmix_packed) -{ - while (nb_samples--) { - *outp[0]++ = inp[0][0]; - *outp[0]++ = inp[0][1]; - inp[0] += aconvert->in_nb_channels; - } -} - -REMATRIX_FUNC_SIG(mono_to_stereo_packed) -{ - while (nb_samples >= 4) { - outp[0][0] = outp[0][1] = inp[0][0]; - outp[0][2] = outp[0][3] = inp[0][1]; - outp[0][4] = outp[0][5] = inp[0][2]; - outp[0][6] = outp[0][7] = inp[0][3]; - outp[0] += 8; - inp[0] += 4; - nb_samples -= 4; - } - while (nb_samples--) { - outp[0][0] = outp[0][1] = inp[0][0]; - outp[0] += 2; - inp[0] += 1; - } -} - -/** - * This is for when we have more than 2 input channels, need to downmix to mono - * and do not have a conversion formula available. We just use first two input - * channels - left and right. This is a placeholder until more conversion - * functions are written. - */ -REMATRIX_FUNC_SIG(mono_downmix_packed) -{ - while (nb_samples--) { - outp[0][0] = (inp[0][0] + inp[0][1]) DIV2; - inp[0] += aconvert->in_nb_channels; - outp[0]++; - } -} - -REMATRIX_FUNC_SIG(mono_downmix_planar) -{ - FMT_TYPE *out = outp[0]; - - while (nb_samples >= 4) { - out[0] = (inp[0][0] + inp[1][0]) DIV2; - out[1] = (inp[0][1] + inp[1][1]) DIV2; - out[2] = (inp[0][2] + inp[1][2]) DIV2; - out[3] = (inp[0][3] + inp[1][3]) DIV2; - out += 4; - inp[0] += 4; - inp[1] += 4; - nb_samples -= 4; - } - while (nb_samples--) { - out[0] = (inp[0][0] + inp[1][0]) DIV2; - out++; - inp[0]++; - inp[1]++; - } -} - -/* Stereo to 5.1 output */ -REMATRIX_FUNC_SIG(stereo_to_surround_5p1_packed) -{ - while (nb_samples--) { - outp[0][0] = inp[0][0]; /* left */ - outp[0][1] = inp[0][1]; /* right */ - outp[0][2] = (inp[0][0] + inp[0][1]) DIV2; /* center */ - outp[0][3] = 0; /* low freq */ - outp[0][4] = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */ - outp[0][5] = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */ - inp[0] += 2; - outp[0] += 6; - } -} - -REMATRIX_FUNC_SIG(stereo_to_surround_5p1_planar) -{ - while (nb_samples--) { - *outp[0]++ = *inp[0]; /* left */ - *outp[1]++ = *inp[1]; /* right */ - *outp[2]++ = (*inp[0] + *inp[1]) DIV2; /* center */ - *outp[3]++ = 0; /* low freq */ - *outp[4]++ = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */ - *outp[5]++ = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */ - inp[0]++; inp[1]++; - } -} - - -/* -5.1 to stereo input: [fl, fr, c, lfe, rl, rr] -- Left = front_left + rear_gain * rear_left + center_gain * center -- Right = front_right + rear_gain * rear_right + center_gain * center -Where rear_gain is usually around 0.5-1.0 and - center_gain is almost always 0.7 (-3 dB) -*/ -REMATRIX_FUNC_SIG(surround_5p1_to_stereo_packed) -{ - while (nb_samples--) { - *outp[0]++ = inp[0][0] + (0.5 * inp[0][4]) + (0.7 * inp[0][2]); //FIXME CLIPPING! - *outp[0]++ = inp[0][1] + (0.5 * inp[0][5]) + (0.7 * inp[0][2]); //FIXME CLIPPING! - - inp[0] += 6; - } -} - -REMATRIX_FUNC_SIG(surround_5p1_to_stereo_planar) -{ - while (nb_samples--) { - *outp[0]++ = *inp[0] + (0.5 * *inp[4]) + (0.7 * *inp[2]); //FIXME CLIPPING! - *outp[1]++ = *inp[1] + (0.5 * *inp[5]) + (0.7 * *inp[2]); //FIXME CLIPPING! - - inp[0]++; inp[1]++; inp[2]++; inp[3]++; inp[4]++; inp[5]++; - } -} - -#undef DIV2 -#undef REMATRIX_FUNC_NAME -#undef FMT_TYPE diff --git a/libavfilter/version.h b/libavfilter/version.h index 393fb918a1..11e038d628 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -29,7 +29,7 @@ #include "libavutil/avutil.h" #define LIBAVFILTER_VERSION_MAJOR 2 -#define LIBAVFILTER_VERSION_MINOR 60 +#define LIBAVFILTER_VERSION_MINOR 61 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \