Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

lavfi/aresample: use libswresample.

Browse Source
pull/3/merge
Clément Bœsch 13 years ago
parent 9f0b0db0d3
commit e96be8409f
3 changed files with 32 additions and 253 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 1
      configure
  2. 2
      libavfilter/Makefile
  3. 282
      libavfilter/af_aresample.c

1
configure vendored
Unescape View File

@ -1649,6 +1649,7 @@ udp_protocol_deps="network"
# filters
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
blackframe_filter_deps="gpl"
boxblur_filter_deps="gpl"

2
libavfilter/Makefile
Unescape View File

@ -5,7 +5,7 @@ FFLIBS = avutil
FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_PAN_FILTER) += swresample
FFLIBS-$(CONFIG_SCALE_FILTER) += swscale

282
libavfilter/af_aresample.c
Unescape View File

@ -24,20 +24,14 @@
* resampling audio filter
*/
#include "libavutil/eval.h"
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
struct AVResampleContext *resample;
int out_rate;
double ratio;
AVFilterBufferRef *outsamplesref;
int unconsumed_nb_samples,
max_cached_nb_samples;
int16_t *cached_data[8],
*resampled_data[8];
struct SwrContext *swr;
} AResampleContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
@ -58,23 +52,12 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
static av_cold void uninit(AVFilterContext *ctx)
{
AResampleContext *aresample = ctx->priv;
if (aresample->outsamplesref) {
int nb_channels =
av_get_channel_layout_nb_channels(
aresample->outsamplesref->audio->channel_layout);
avfilter_unref_buffer(aresample->outsamplesref);
while (nb_channels--) {
av_freep(&(aresample->cached_data[nb_channels]));
av_freep(&(aresample->resampled_data[nb_channels]));
}
}
if (aresample->resample)
av_resample_close(aresample->resample);
swr_free(&aresample->swr);
}
static int config_output(AVFilterLink *outlink)
{
int ret;
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AResampleContext *aresample = ctx->priv;
@ -85,9 +68,16 @@ static int config_output(AVFilterLink *outlink)
outlink->sample_rate = aresample->out_rate;
outlink->time_base = (AVRational) {1, aresample->out_rate};
//TODO: make the resampling parameters configurable
aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
16, 10, 0, 0.8);
//TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
aresample->swr = swr_alloc_set_opts(aresample->swr,
inlink->channel_layout, inlink->format, aresample->out_rate,
inlink->channel_layout, inlink->format, inlink->sample_rate,
0, ctx);
if (!aresample->swr)
return AVERROR(ENOMEM);
ret = swr_init(aresample->swr);
if (ret < 0)
return ret;
aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
@ -96,235 +86,24 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_sample_formats(ctx, formats);
formats = avfilter_make_all_channel_layouts();
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_channel_layouts(ctx, formats);
formats = avfilter_make_all_packing_formats();
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_packing_formats(ctx, formats);
return 0;
}
static void deinterleave(int16_t **outp, int16_t *in,
int nb_channels, int nb_samples)
{
int16_t *out[8];
memcpy(out, outp, nb_channels * sizeof(int16_t*));
switch (nb_channels) {
case 2:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
}
break;
case 3:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
*out[2]++ = *in++;
}
break;
case 4:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
*out[2]++ = *in++;
*out[3]++ = *in++;
}
break;
case 5:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
*out[2]++ = *in++;
*out[3]++ = *in++;
*out[4]++ = *in++;
}
break;
case 6:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
*out[2]++ = *in++;
*out[3]++ = *in++;
*out[4]++ = *in++;
*out[5]++ = *in++;
}
break;
case 8:
while (nb_samples--) {
*out[0]++ = *in++;
*out[1]++ = *in++;
*out[2]++ = *in++;
*out[3]++ = *in++;
*out[4]++ = *in++;
*out[5]++ = *in++;
*out[6]++ = *in++;
*out[7]++ = *in++;
}
break;
}
}
static void interleave(int16_t *out, int16_t **inp,
int nb_channels, int nb_samples)
{
int16_t *in[8];
memcpy(in, inp, nb_channels * sizeof(int16_t*));
switch (nb_channels) {
case 2:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
}
break;
case 3:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
*out++ = *in[2]++;
}
break;
case 4:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
*out++ = *in[2]++;
*out++ = *in[3]++;
}
break;
case 5:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
*out++ = *in[2]++;
*out++ = *in[3]++;
*out++ = *in[4]++;
}
break;
case 6:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
*out++ = *in[2]++;
*out++ = *in[3]++;
*out++ = *in[4]++;
*out++ = *in[5]++;
}
break;
case 8:
while (nb_samples--) {
*out++ = *in[0]++;
*out++ = *in[1]++;
*out++ = *in[2]++;
*out++ = *in[3]++;
*out++ = *in[4]++;
*out++ = *in[5]++;
*out++ = *in[6]++;
*out++ = *in[7]++;
}
break;
}
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int i,
in_nb_samples = insamplesref->audio->nb_samples,
cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
requested_out_nb_samples = aresample->ratio * cached_nb_samples,
nb_channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
if (cached_nb_samples > aresample->max_cached_nb_samples) {
for (i = 0; i < nb_channels; i++) {
aresample->cached_data[i] =
av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
aresample->resampled_data[i] =
av_realloc(aresample->resampled_data[i],
FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
return;
}
aresample->max_cached_nb_samples = cached_nb_samples;
if (aresample->outsamplesref)
avfilter_unref_buffer(aresample->outsamplesref);
aresample->outsamplesref =
avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
outlink->out_buf = aresample->outsamplesref;
}
avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
aresample->outsamplesref->pts =
av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
/* av_resample() works with planar audio buffers */
if (!inlink->planar && nb_channels > 1) {
int16_t *out[8];
for (i = 0; i < nb_channels; i++)
out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
deinterleave(out, (int16_t *)insamplesref->data[0],
nb_channels, in_nb_samples);
} else {
for (i = 0; i < nb_channels; i++)
memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
insamplesref->data[i],
in_nb_samples * sizeof(int16_t));
}
for (i = 0; i < nb_channels; i++) {
int consumed_nb_samples;
const int is_last = i+1 == nb_channels;
aresample->outsamplesref->audio->nb_samples =
av_resample(aresample->resample,
aresample->resampled_data[i], aresample->cached_data[i],
&consumed_nb_samples,
cached_nb_samples,
requested_out_nb_samples, is_last);
/* move unconsumed data back to the beginning of the cache */
aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
memmove(aresample->cached_data[i],
aresample->cached_data[i] + consumed_nb_samples,
aresample->unconsumed_nb_samples * sizeof(int16_t));
}
/* copy resampled data to the output samplesref */
if (!inlink->planar && nb_channels > 1) {
interleave((int16_t *)aresample->outsamplesref->data[0],
aresample->resampled_data,
nb_channels, aresample->outsamplesref->audio->nb_samples);
} else {
for (i = 0; i < nb_channels; i++)
memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
}
avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
(void *)insamplesref->data, n_in);
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
outsamplesref->pts = av_rescale(outlink->sample_rate, insamplesref->pts,
inlink ->sample_rate);
avfilter_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
}
@ -333,7 +112,6 @@ AVFilter avfilter_af_aresample = {
.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.priv_size = sizeof(AResampleContext),
.inputs = (const AVFilterPad[]) {{ .name = "default",

Write Preview
Loading…
Cancel
Save