[FFmpeg-devel] [PATCH 3/4] lavfi: add audio convert filter
Mina Nagy Zaki
mnzaki at gmail.com
Wed Jul 27 00:05:19 CEST 2011
Add aconvert filter to perform sample format and channel layout conversion.
Based on code by Stefano Sabatini and "S.N. Hemanth Meenakshisundaram"
smeenaks at ucsd.edu.
---
libavfilter/Makefile | 1 +
libavfilter/af_aconvert.c | 430 ++++++++++++++++++++++++++++++++++++
libavfilter/af_aconvert_rematrix.c | 185 ++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 617 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/af_aconvert.c
create mode 100644 libavfilter/af_aconvert_rematrix.c
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 83b906d..0e6051b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -18,6 +18,7 @@ OBJS = allfilters.o \
OBJS-$(CONFIG_AVCODEC) += avcodec.o
+OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
diff --git a/libavfilter/af_aconvert.c b/libavfilter/af_aconvert.c
new file mode 100644
index 0000000..0f8ba33
--- /dev/null
+++ b/libavfilter/af_aconvert.c
@@ -0,0 +1,430 @@
+/*
+ * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks at ucsd.edu>
+ * Copyright (c) 2011 Stefano Sabatini
+ * 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
+ * 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 "avfilter.h"
+#include "libavcodec/audioconvert.h"
+
+#define SFMT_t uint8_t
+#define REMATRIX(FUNC) FUNC ## _u8
+#include "af_aconvert_rematrix.c"
+
+#define SFMT_t int16_t
+#define REMATRIX(FUNC) FUNC ## _s16
+#include "af_aconvert_rematrix.c"
+
+#define SFMT_t int32_t
+#define REMATRIX(FUNC) FUNC ## _s32
+#include "af_aconvert_rematrix.c"
+
+#define FLOATING
+
+#define SFMT_t float
+#define REMATRIX(FUNC) FUNC ## _flt
+#include "af_aconvert_rematrix.c"
+
+#define SFMT_t double
+#define REMATRIX(FUNC) FUNC ## _dbl
+#include "af_aconvert_rematrix.c"
+
+typedef struct {
+ int nb_samples; ///< current size of buffers
+ enum AVSampleFormat out_sample_fmt; ///< output sample format
+ int64_t out_chlayout; ///< output channel layout
+
+ int out_strides[8],
+ in_strides [8];
+
+ AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer
+ AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions
+ uint8_t *packed_data[8]; ///< pointers for packing conversion
+ uint8_t **in_data, **out_data; ///< input/output for av_audio_convert
+
+ AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format
+
+ void (*convert_chlayout) ();
+} AConvertContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ AConvertContext *aconvert = ctx->priv;
+ char sample_fmt_str[8] = "", chlayout_str[32] = "";
+
+ if (args)
+ sscanf(args, "%8[^:]:%32s", sample_fmt_str, chlayout_str);
+
+ aconvert->out_sample_fmt =
+ *sample_fmt_str ? av_get_sample_fmt(sample_fmt_str) : AV_SAMPLE_FMT_NONE;
+
+ if (*sample_fmt_str && aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE) {
+ /* -1 is a valid value for out_sample_fmt and indicates no change
+ * in sample format. */
+ char *tail;
+ aconvert->out_sample_fmt = strtol(sample_fmt_str, &tail, 10);
+ if (*tail || (aconvert->out_sample_fmt >= AV_SAMPLE_FMT_NB &&
+ aconvert->out_sample_fmt != -1)) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid sample format '%s'\n",
+ sample_fmt_str);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ aconvert->out_chlayout = *chlayout_str ?
+ av_get_channel_layout(chlayout_str) : -1;
+
+ if (*chlayout_str && aconvert->out_chlayout < AV_CH_LAYOUT_STEREO) {
+ /* -1 is a valid value for out_chlayout and indicates no change
+ * in channel layout. */
+ char *tail;
+ aconvert->out_chlayout = strtol(chlayout_str, &tail, 10);
+ if (*tail || (aconvert->out_chlayout < AV_CH_LAYOUT_STEREO &&
+ aconvert->out_chlayout != -1)) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid channel layout %s\n",
+ chlayout_str);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ return 0;
+}
+
+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);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AConvertContext *aconvert = ctx->priv;
+
+ avfilter_formats_ref(avfilter_all_packing_formats(),
+ &ctx->outputs[0]->in_packing);
+ avfilter_formats_ref(avfilter_all_packing_formats(),
+ &ctx->inputs[0]->out_packing);
+
+ avfilter_formats_ref(avfilter_all_formats(AVMEDIA_TYPE_AUDIO),
+ &ctx->inputs[0]->out_formats);
+ if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
+ avfilter_add_format(&formats, aconvert->out_sample_fmt);
+ avfilter_formats_ref(formats, &ctx->outputs[0]->in_formats);
+ } else
+ avfilter_formats_ref(avfilter_all_formats(AVMEDIA_TYPE_AUDIO),
+ &ctx->outputs[0]->in_formats);
+
+ avfilter_formats_ref(avfilter_all_channel_layouts(),
+ &ctx->inputs[0]->out_chlayouts);
+ if (aconvert->out_chlayout != -1) {
+ formats = NULL;
+ avfilter_add_format(&formats, aconvert->out_chlayout);
+ avfilter_formats_ref(formats, &ctx->outputs[0]->in_chlayouts);
+ } else
+ avfilter_formats_ref(avfilter_all_channel_layouts(),
+ &ctx->outputs[0]->in_chlayouts);
+
+ return 0;
+}
+
+#define CHOOSE_FUNC_SFMT(FUNC) \
+ switch (inlink->format) { \
+ case AV_SAMPLE_FMT_U8: \
+ aconvert->convert_chlayout = FUNC ## _u8; break; \
+ case AV_SAMPLE_FMT_S16: \
+ aconvert->convert_chlayout = FUNC ## _s16; break; \
+ case AV_SAMPLE_FMT_S32: \
+ aconvert->convert_chlayout = FUNC ## _s32; break; \
+ case AV_SAMPLE_FMT_FLT: \
+ aconvert->convert_chlayout = FUNC ## _flt; break; \
+ case AV_SAMPLE_FMT_DBL: \
+ aconvert->convert_chlayout = FUNC ## _dbl; break; \
+ }
+
+#define CHOOSE_FUNC(OUT, FUNC) \
+ if (aconvert->out_chlayout == OUT) { \
+ if (inlink->planar) \
+ CHOOSE_FUNC_SFMT(FUNC ## _planar) \
+ else \
+ CHOOSE_FUNC_SFMT(FUNC ## _packed) \
+ }
+
+#define CHOOSE_FUNC2(IN, OUT, FUNC) \
+ if (inlink->channel_layout == IN && \
+ aconvert->out_chlayout == OUT) { \
+ if (inlink->planar) \
+ CHOOSE_FUNC_SFMT(FUNC ## _planar) \
+ else \
+ CHOOSE_FUNC_SFMT(FUNC ## _packed) \
+ }
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterLink *inlink = outlink->src->inputs[0];
+ AConvertContext *aconvert = outlink->src->priv;
+ char buf1[32], buf2[32];
+
+ /* 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 == -1)
+ aconvert->out_chlayout = outlink->channel_layout;
+
+ 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, "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);
+
+ /* handle stereo_to_mono and mono_to_stereo separately because there are
+ * no planar versions */
+ if (!inlink->planar &&
+ inlink->channel_layout == AV_CH_LAYOUT_STEREO &&
+ aconvert->out_chlayout == AV_CH_LAYOUT_MONO) {
+ CHOOSE_FUNC_SFMT(stereo_to_mono_packed);
+ }
+ else
+ if (!outlink->planar &&
+ inlink->channel_layout == AV_CH_LAYOUT_MONO &&
+ aconvert->out_chlayout == AV_CH_LAYOUT_STEREO) {
+ CHOOSE_FUNC_SFMT(mono_to_stereo_packed);
+ }
+
+ if (!aconvert->convert_chlayout &&
+ inlink->channel_layout != outlink->channel_layout) {
+ CHOOSE_FUNC2(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, ac3_5p1_mux)
+ else CHOOSE_FUNC2(AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_to_stereo)
+ else CHOOSE_FUNC( AV_CH_LAYOUT_MONO, mono_downmix)
+ }
+
+ /* If there's no channel conversion function and output is stereo,
+ * we can do generic stereo downmixing:
+ * if there's a format conversion then stereo downmixing is implicitly
+ * done by av_audio_convert.
+ * if there's no format conversion then packed stereo downmixing is
+ * explicitly done by av_audio_convert, while planar is done in
+ * filter_samples
+ */
+ if (!aconvert->convert_chlayout &&
+ outlink->channel_layout != inlink->channel_layout &&
+ outlink->channel_layout != AV_CH_LAYOUT_STEREO) {
+ av_log(outlink->src, AV_LOG_ERROR,
+ "Unsupported channel layout conversion requested!\n");
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+static void init_buffers(AVFilterLink *inlink, int nb_samples)
+{
+ AConvertContext *aconvert = inlink->dst->priv;
+ AVFilterLink * const outlink = inlink->dst->outputs[0];
+ int i, packed_stride = 0;
+ int in_channels =
+ av_get_channel_layout_nb_channels(inlink->channel_layout),
+ out_channels =
+ av_get_channel_layout_nb_channels(outlink->channel_layout);
+ const short
+ stereo_downmix = inlink->channel_layout != outlink->channel_layout &&
+ !aconvert->convert_chlayout,
+ format_conv = inlink->format != outlink->format,
+ packing_conv = inlink->planar != outlink->planar &&
+ in_channels != 1 &&
+ out_channels != 1;
+
+ aconvert->nb_samples = nb_samples;
+ uninit(inlink->dst);
+
+ // rematrixing
+ if (aconvert->convert_chlayout) {
+ aconvert->mix_samplesref =
+ avfilter_get_audio_buffer(outlink,
+ AV_PERM_WRITE | AV_PERM_REUSE2,
+ inlink->format,
+ nb_samples,
+ outlink->channel_layout,
+ inlink->planar);
+ in_channels = out_channels;
+ }
+
+ /* If there's any conversion left to do, we need a buffer */
+ if (format_conv || packing_conv || stereo_downmix) {
+ aconvert->out_samplesref = avfilter_get_audio_buffer(outlink,
+ AV_PERM_WRITE | AV_PERM_REUSE2,
+ outlink->format,
+ nb_samples,
+ outlink->channel_layout,
+ outlink->planar);
+ }
+
+ /* if there's a format/mode conversion or packed stereo downmixing,
+ * we need an audio_convert context
+ */
+ if (format_conv || packing_conv || (stereo_downmix && !outlink->planar)) {
+ aconvert->in_strides[0] = av_get_bytes_per_sample(inlink->format);
+ aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format);
+
+ aconvert->out_data = aconvert->out_samplesref->data;
+ if (aconvert->mix_samplesref)
+ aconvert->in_data = aconvert->mix_samplesref->data;
+
+ if (packing_conv) {
+ if (outlink->planar) {
+ if (aconvert->mix_samplesref)
+ aconvert->packed_data[0] =
+ aconvert->mix_samplesref->data[0];
+ aconvert->in_data = aconvert->packed_data;
+ packed_stride = aconvert->in_strides[0];
+ aconvert->in_strides[0] *= in_channels;
+ } else {
+ aconvert->packed_data[0] = aconvert->out_samplesref->data[0];
+ aconvert->out_data = aconvert->packed_data;
+ packed_stride = aconvert->out_strides[0];
+ aconvert->out_strides[0] *= out_channels;
+ }
+ } else if (!outlink->planar) {
+ out_channels = 1;
+ }
+
+ for (i = 1; i < out_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, out_channels,
+ inlink->format, out_channels, NULL, 0);
+
+ }
+
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+{
+ AConvertContext *aconvert = inlink->dst->priv;
+ AVFilterBufferRef *curbuf = insamplesref;
+ AVFilterLink * const outlink = inlink->dst->outputs[0];
+ int nb_channels = av_get_channel_layout_nb_channels(
+ curbuf->audio->channel_layout);
+
+ if (!aconvert->nb_samples ||
+ (curbuf->audio->nb_samples > aconvert->nb_samples))
+ init_buffers(inlink, curbuf->audio->nb_samples);
+
+ if (aconvert->mix_samplesref) {
+ if (inlink->planar && nb_channels != 1)
+ aconvert->convert_chlayout(aconvert->mix_samplesref->data,
+ curbuf->data,
+ curbuf->audio->nb_samples,
+ nb_channels);
+ else
+ aconvert->convert_chlayout(aconvert->mix_samplesref->data[0],
+ curbuf->data[0],
+ curbuf->audio->nb_samples,
+ nb_channels);
+
+ aconvert->mix_samplesref->audio->nb_samples =
+ curbuf->audio->nb_samples;
+ curbuf = aconvert->mix_samplesref;
+
+ }
+
+ if (aconvert->audioconvert_ctx) {
+ if (!aconvert->mix_samplesref) {
+ if (aconvert->in_data == 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 < nb_channels; i++)
+ aconvert->packed_data[i] =
+ aconvert->packed_data[i-1] + packed_stride;
+ } else {
+ aconvert->in_data = curbuf->data;
+ }
+ }
+
+ if (inlink->planar == outlink->planar && !outlink->planar)
+ nb_channels = av_get_channel_layout_nb_channels(
+ curbuf->audio->channel_layout);
+ else
+ nb_channels = 1;
+
+ av_audio_convert(aconvert->audioconvert_ctx,
+ (void * const *) aconvert->out_data,
+ aconvert->out_strides,
+ (const void * const *) aconvert->in_data,
+ aconvert->in_strides,
+ curbuf->audio->nb_samples * nb_channels);
+
+ aconvert->out_samplesref->audio->nb_samples =
+ curbuf->audio->nb_samples;
+ curbuf = aconvert->out_samplesref;
+ }
+
+ /* Handle generic planar stereo downmixing */
+ if (!aconvert->convert_chlayout && !aconvert->audioconvert_ctx &&
+ outlink->channel_layout == AV_CH_LAYOUT_STEREO) {
+ int size =
+ av_get_bytes_per_sample(inlink->format) * curbuf->audio->nb_samples;
+ if (nb_channels == 1) curbuf->data[1] = curbuf->data[0];
+ memcpy(aconvert->out_samplesref->data[0],curbuf->data[0], size);
+ memcpy(aconvert->out_samplesref->data[1], curbuf->data[1], size);
+ aconvert->out_samplesref->audio->nb_samples =
+ curbuf->audio->nb_samples;
+ curbuf = aconvert->out_samplesref;
+ }
+
+ avfilter_filter_samples(inlink->dst->outputs[0],
+ avfilter_ref_buffer(curbuf, ~0));
+ avfilter_unref_buffer(insamplesref);
+}
+
+AVFilter avfilter_af_aconvert = {
+ .name = "aconvert",
+ .description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout."),
+ .priv_size = sizeof(AConvertContext),
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+
+ .inputs = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ, },
+ { .name = NULL}},
+ .outputs = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output, },
+ { .name = NULL}},
+};
diff --git a/libavfilter/af_aconvert_rematrix.c b/libavfilter/af_aconvert_rematrix.c
new file mode 100644
index 0000000..3e538e7
--- /dev/null
+++ b/libavfilter/af_aconvert_rematrix.c
@@ -0,0 +1,185 @@
+/*
+ * 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
+ */
+
+#if defined(FLOATING)
+# define DIV2 /2
+#else
+# define DIV2 >>1
+#endif
+
+#define REMATRIX_FUNC_PACKED(FUNC) static void REMATRIX(FUNC) \
+ (SFMT_t *out, SFMT_t *in, int nb_samples, int in_channels)
+#define REMATRIX_FUNC_PLANAR(FUNC) static void REMATRIX(FUNC) \
+ (SFMT_t *outp[], SFMT_t *inp[], int nb_samples, int in_channels)
+
+REMATRIX_FUNC_PACKED(stereo_to_mono_packed)
+{
+ while (nb_samples >= 4) {
+ out[0] = (in[0] + in[1]) DIV2;
+ out[1] = (in[2] + in[3]) DIV2;
+ out[2] = (in[4] + in[5]) DIV2;
+ out[3] = (in[6] + in[7]) DIV2;
+ out += 4;
+ in += 8;
+ nb_samples -= 4;
+ }
+ while (nb_samples--) {
+ out[0] = (in[0] + in[1]) DIV2;
+ out++;
+ in += 2;
+ }
+}
+
+REMATRIX_FUNC_PACKED(mono_to_stereo_packed)
+{
+ while (nb_samples >= 4) {
+ out[0] = out[1] = in[0];
+ out[2] = out[3] = in[1];
+ out[4] = out[5] = in[2];
+ out[6] = out[7] = in[3];
+ out += 8;
+ in += 4;
+ nb_samples -= 4;
+ }
+ while (nb_samples--) {
+ out[0] = out[1] = in[0];
+ out += 2;
+ in += 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_PACKED(mono_downmix_packed)
+{
+ while (nb_samples--) {
+ out[0] = (in[0] + in[1]) DIV2;
+ in += in_channels;
+ out++;
+ }
+}
+
+REMATRIX_FUNC_PLANAR(mono_downmix_planar)
+{
+ SFMT_t *in[2], *out = outp[0];
+ in[0] = inp[0];
+ in[1] = inp[1];
+
+ while (nb_samples >= 4) {
+ out[0] = (in[0][0] + in[1][0]) DIV2;
+ out[1] = (in[0][1] + in[1][1]) DIV2;
+ out[2] = (in[0][2] + in[1][2]) DIV2;
+ out[3] = (in[0][3] + in[1][3]) DIV2;
+ out += 4;
+ in[0] += 4;
+ in[1] += 4;
+ nb_samples -= 4;
+ }
+ while (nb_samples--) {
+ out[0] = (in[0][0] + in[1][0]) DIV2;
+ out++;
+ in[0]++;
+ in[1]++;
+ }
+
+}
+
+/* Stereo to 5.1 output */
+REMATRIX_FUNC_PACKED(ac3_5p1_mux_packed)
+{
+ while (nb_samples--) {
+ out[0] = in[0]; /* left */
+ out[1] = in[1]; /* right */
+ out[2] = (in[0] + in[1]) DIV2; /* center */
+ out[3] = 0; /* low freq */
+ out[4] = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */
+ out[5] = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
+ in += 2;
+ out += 6;
+ }
+}
+
+REMATRIX_FUNC_PLANAR(ac3_5p1_mux_planar)
+{
+ SFMT_t *in[2], *out[6];
+ in[0] = inp[0]; in[1] = inp[1];
+ out[0] = outp[0]; out[1] = outp[1];
+ out[2] = outp[2]; out[3] = outp[3];
+ out[4] = outp[4]; out[5] = outp[5];
+
+ while (nb_samples--) {
+ *out[0]++ = *in[0]; /* left */
+ *out[1]++ = *in[1]; /* right */
+ *out[2]++ = (*in[0] + *in[1]) DIV2; /* center */
+ *out[3]++ = 0; /* low freq */
+ *out[4]++ = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */
+ *out[5]++ = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
+ in[0]++; in[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_PACKED(surround_to_stereo_packed)
+{
+ while (nb_samples--) {
+ *out++ = in[0] + (0.5 * in[4]) + (0.7 * in[2]); //FIXME CLIPPING!
+ *out++ = in[1] + (0.5 * in[5]) + (0.7 * in[2]); //FIXME CLIPPING!
+
+ in += 6;
+ }
+}
+
+REMATRIX_FUNC_PLANAR(surround_to_stereo_planar)
+{
+ SFMT_t *in[6], *out[2];
+ out[0] = outp[0];
+ out[1] = outp[1];
+ in[0] = inp[0]; in[1] = inp[1];
+ in[2] = inp[2]; in[3] = inp[3];
+ in[4] = inp[4]; in[5] = inp[5];
+
+ while (nb_samples--) {
+ *out[0]++ = *in[0] + (0.5 * *in[4]) + (0.7 * *in[2]); //FIXME CLIPPING!
+ *out[1]++ = *in[1] + (0.5 * *in[5]) + (0.7 * *in[2]); //FIXME CLIPPING!
+
+ in[0]++; in[1]++; in[2]++; in[3]++; in[4]++; in[5]++;
+ }
+}
+
+#undef DIV2
+#undef REMATRIX
+#undef REMATRIX_FUNC
+#undef SFMT_t
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index b812ff7..4a4642d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -34,6 +34,7 @@ void avfilter_register_all(void)
return;
initialized = 1;
+ REGISTER_FILTER (ACONVERT, aconvert, af);
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (ANULL, anull, af);
--
1.7.4.4
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