[FFmpeg-devel] [PATCH][WIP] lavfi/aconvert: use libswresample.
Clément Bœsch
ubitux at gmail.com
Thu Jan 26 12:43:36 CET 2012
---
In the same spirit of the previous aresample attempt to use libswresample, here
is the one for aconvert.
And just like aresample, I hit the issue of the duration (ffmpeg -t) being
ignored. Using copy_props made things worse, and copying pts and pos manually
didn't get things better, so I don't know what's wrong here. The duration is
*not* ignored with the current version so there is definitely something wrong
to fix before I can push these patches.
Also, after this patch, this crashes:
./ffplay -f lavfi -i amovie=a.flac,aconvert=s16p:stereo:auto
while this doesn't:
./ffplay -f lavfi -i amovie=a.flac,aconvert=s16p:stereo:0
./ffplay -f lavfi -i amovie=a.flac,aconvert=s16p:stereo:1
Input is s16 stereo packed. I wasn't able to test with current version because
stereo→mono isn't supported. (no idea why it's trying to use mono though).
I think I mess with the way I request the packing, but to avoid the issue I
could just remove the packing parameter, which is now part of the sample
format. OTOH, we won't be able to use "auto" for the packing anymore.
BTW, what is planned for the audio packing in lavfi?
---
configure | 1 +
libavfilter/Makefile | 2 +-
libavfilter/af_aconvert.c | 292 ++++--------------------------------
libavfilter/af_aconvert_rematrix.c | 172 ---------------------
4 files changed, 34 insertions(+), 433 deletions(-)
delete mode 100644 libavfilter/af_aconvert_rematrix.c
diff --git a/configure b/configure
index 6cfadb3..a54f086 100755
--- a/configure
+++ b/configure
@@ -1645,6 +1645,7 @@ 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/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..4cb54cb 100644
--- a/libavfilter/af_aconvert.c
+++ b/libavfilter/af_aconvert.c
@@ -23,98 +23,20 @@
/**
* @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;
+ enum AVFilterPacking out_packing_fmt;
+ 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;
@@ -147,10 +69,7 @@ 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)
@@ -163,6 +82,9 @@ static int query_formats(AVFilterContext *ctx)
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&inlink->out_formats);
if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
+ if (aconvert->out_packing_fmt != -1)
+ aconvert->out_sample_fmt = av_get_alt_sample_fmt(aconvert->out_sample_fmt,
+ aconvert->out_packing_fmt);
formats = NULL;
avfilter_add_format(&formats, aconvert->out_sample_fmt);
avfilter_formats_ref(formats, &outlink->in_formats);
@@ -195,206 +117,56 @@ static int query_formats(AVFilterContext *ctx)
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);
+ aconvert->out_sample_fmt = av_get_alt_sample_fmt(aconvert->out_sample_fmt,
+ aconvert->out_packing_fmt);
/* 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;
- }
-
- /* 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;
- }
+ 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);
- avfilter_copy_buffer_ref_props(curbuf, insamplesref);
- curbuf->audio->channel_layout = outlink->channel_layout;
- curbuf->audio->planar = outlink->planar;
+ swr_convert(aconvert->swr, outsamplesref->data, n,
+ (void *)insamplesref->data, n);
- 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
--
1.7.8.4
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