mirror of https://github.com/FFmpeg/FFmpeg.git
Add aconvert filter to perform sample format, channel layout, and packing format conversion. The aconvert code depends on audio conversion code in libavcodec, so this requires a dependency on libavcodec. Based on previous work by S.N. Hemanth Meenakshisundaram and Mina Nagy Zaki, performed for the GSoC 2010 and 2011.pull/2/head
Stefano Sabatini
14 years ago
committed by
Stefano Sabatini
parent
553c5e9f23
commit
37cc443c83
8 changed files with 632 additions and 1 deletions
@ -0,0 +1,417 @@ |
||||
/*
|
||||
* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@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 "libavutil/audioconvert.h" |
||||
#include "libavcodec/audioconvert.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
|
||||
} 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 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; |
||||
char *arg, *ptr = NULL; |
||||
int ret = 0; |
||||
char *args = av_strdup(args0); |
||||
|
||||
aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE; |
||||
aconvert->out_chlayout = 0; |
||||
aconvert->out_packing_fmt = -1; |
||||
|
||||
if ((arg = strtok_r(args, ":", &ptr)) && strcmp(arg, "auto")) { |
||||
if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0) |
||||
goto end; |
||||
} |
||||
if ((arg = strtok_r(NULL, ":", &ptr)) && strcmp(arg, "auto")) { |
||||
if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0) |
||||
goto end; |
||||
} |
||||
if ((arg = strtok_r(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); |
||||
return ret; |
||||
} |
||||
|
||||
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; |
||||
AVFilterLink *inlink = ctx->inputs[0]; |
||||
AVFilterLink *outlink = ctx->outputs[0]; |
||||
|
||||
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO), |
||||
&inlink->out_formats); |
||||
if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) { |
||||
formats = NULL; |
||||
avfilter_add_format(&formats, aconvert->out_sample_fmt); |
||||
avfilter_formats_ref(formats, &outlink->in_formats); |
||||
} else |
||||
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO), |
||||
&outlink->in_formats); |
||||
|
||||
avfilter_formats_ref(avfilter_make_all_channel_layouts(), |
||||
&inlink->out_chlayouts); |
||||
if (aconvert->out_chlayout != 0) { |
||||
formats = NULL; |
||||
avfilter_add_format(&formats, aconvert->out_chlayout); |
||||
avfilter_formats_ref(formats, &outlink->in_chlayouts); |
||||
} else |
||||
avfilter_formats_ref(avfilter_make_all_channel_layouts(), |
||||
&outlink->in_chlayouts); |
||||
|
||||
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); |
||||
|
||||
return 0; |
||||
} |
||||
|
||||
static int config_output(AVFilterLink *outlink) |
||||
{ |
||||
AVFilterLink *inlink = outlink->src->inputs[0]; |
||||
AConvertContext *aconvert = outlink->src->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); |
||||
|
||||
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); |
||||
|
||||
/* 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, inlink->planar, 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; |
||||
} |
||||
|
||||
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_unref_buffer(insamplesref); |
||||
} |
||||
|
||||
AVFilter avfilter_af_aconvert = { |
||||
.name = "aconvert", |
||||
.description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout:packed_fmt."), |
||||
.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}}, |
||||
}; |
||||
@ -0,0 +1,172 @@ |
||||
/*
|
||||
* 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 |
||||
Loading…
Reference in new issue