[FFmpeg-devel] [PATCH 3/4] lavfi/aconvert: use libswresample.
Stefano Sabatini
stefasab at gmail.com
Tue Jan 31 16:08:48 CET 2012
On date Tuesday 2012-01-31 08:31:47 +0100, Clément Bœsch encoded:
> This commit also drops the planar parameter; you now need to use the 'p'
> suffix in order to request a planar sample format.
> ---
> configure | 1 +
> doc/filters.texi | 19 +--
> libavfilter/Makefile | 2 +-
> libavfilter/af_aconvert.c | 307 ++++--------------------------------
> libavfilter/af_aconvert_rematrix.c | 172 --------------------
> libavfilter/version.h | 2 +-
> 6 files changed, 44 insertions(+), 459 deletions(-)
> delete mode 100644 libavfilter/af_aconvert_rematrix.c
>
> diff --git a/configure b/configure
> index b4e433c..aa45fd7 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 7d008bc..ef96910 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}".
>
> - at var{sample_format} specifies the sample format, and can be a string or
> -the corresponding numeric value defined in @file{libavutil/samplefmt.h}.
> + at 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}.
>
> - at 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 ff0ba75..9fbb59b 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 e3c7f8c..8c1b5dc 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;
> -
> - /* 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;
> - }
> + 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);
>
> - /* 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;
> - }
> -
> - 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;
> - }
> + swr_convert(aconvert->swr, outsamplesref->data, n,
> + (void *)insamplesref->data, n);
>
> - avfilter_copy_buffer_ref_props(curbuf, insamplesref);
> - curbuf->audio->channel_layout = outlink->channel_layout;
> - curbuf->audio->planar = outlink->planar;
> + avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
> + outsamplesref->audio->channel_layout = outlink->channel_layout;
> + outsamplesref->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 d75ca5a..0000000
> --- 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 60e496d..cd8bd95 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, \
Sounds good, I suppose some minimal tests have been done on it, a
complete FATE test is definitively welcome at this point.
--
FFmpeg = Fabulous and Fundamental Mystic Pitiful Erotic Game
More information about the ffmpeg-devel
mailing list