Skip to content

F
ffmpeg.wasm-core
Commit d4ce63a1 authored by Paul B Mahol's avatar Paul B Mahol
Browse files
Options

avfilter/af_sidechaincompress & af_agate: use audio fifo from lavu 

Fixes regression causing segfault.
Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent eded2e4f
Hide whitespace changes
Inline Side-by-side
Showing with 102 additions and 55 deletions
libavfilter/af_agate.c
View file @ d4ce63a1
......@@ -23,6 +23,7 @@
* Audio (Sidechain) Gate filter
*/
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
......@@ -54,7 +55,8 @@ typedef struct AudioGateContext {
double attack_coeff;
double release_coeff;
AVFrame *input_frame[2];
AVAudioFifo *fifo[2];
int64_t pts;
} AudioGateContext;
#define OFFSET(x) offsetof(AudioGateContext, x)
......@@ -263,58 +265,67 @@ AVFilter ff_af_agate = {
#define sidechaingate_options options
AVFILTER_DEFINE_CLASS(sidechaingate);
static int scfilter_frame(AVFilterLink *link, AVFrame *in)
static int scfilter_frame(AVFilterLink *link, AVFrame *frame)
{
AVFilterContext *ctx = link->dst;
AudioGateContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
const double *scsrc;
double *sample;
AVFrame *out, *in[2] = { NULL };
double *dst;
int nb_samples;
int ret, i;
int i;
for (i = 0; i < 2; i++)
if (link == ctx->inputs[i])
break;
av_assert0(i < 2 && !s->input_frame[i]);
s->input_frame[i] = in;
av_assert0(i < 2);
av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
frame->nb_samples);
av_frame_free(&frame);
if (!s->input_frame[0] || !s->input_frame[1])
nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
if (!nb_samples)
return 0;
nb_samples = FFMIN(s->input_frame[0]->nb_samples,
s->input_frame[1]->nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);
for (i = 0; i < 2; i++) {
in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
if (!in[i]) {
av_frame_free(&in[0]);
av_frame_free(&in[1]);
return AVERROR(ENOMEM);
}
av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
}
sample = (double *)s->input_frame[0]->data[0];
scsrc = (const double *)s->input_frame[1]->data[0];
dst = (double *)out->data[0];
out->pts = s->pts;
s->pts += nb_samples;
gate(s, sample, sample, scsrc, nb_samples,
gate(s, (double *)in[0]->data[0], dst,
(double *)in[1]->data[0], nb_samples,
s->level_in, s->level_sc,
ctx->inputs[0], ctx->inputs[1]);
ret = ff_filter_frame(outlink, s->input_frame[0]);
s->input_frame[0] = NULL;
av_frame_free(&s->input_frame[1]);
av_frame_free(&in[0]);
av_frame_free(&in[1]);
return ret;
return ff_filter_frame(outlink, out);
}
static int screquest_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioGateContext *s = ctx->priv;
int i, ret;
int i;
/* get a frame on each input */
for (i = 0; i < 2; i++) {
AVFilterLink *inlink = ctx->inputs[i];
if (!s->input_frame[i] &&
(ret = ff_request_frame(inlink)) < 0)
return ret;
/* request the same number of samples on all inputs */
if (i == 0)
ctx->inputs[1]->request_samples = s->input_frame[0]->nb_samples;
if (!av_audio_fifo_size(s->fifo[i]))
return ff_request_frame(inlink);
}
return 0;
......@@ -358,6 +369,7 @@ static int scquery_formats(AVFilterContext *ctx)
static int scconfig_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioGateContext *s = ctx->priv;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
......@@ -372,23 +384,34 @@ static int scconfig_output(AVFilterLink *outlink)
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
if (!s->fifo[0] || !s->fifo[1])
return AVERROR(ENOMEM);
agate_config_input(ctx->inputs[0]);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioGateContext *s = ctx->priv;
av_audio_fifo_free(s->fifo[0]);
av_audio_fifo_free(s->fifo[1]);
}
static const AVFilterPad sidechaingate_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = scfilter_frame,
.needs_writable = 1,
.needs_fifo = 1,
},{
.name = "sidechain",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = scfilter_frame,
.needs_fifo = 1,
},
{ NULL }
};
......@@ -409,6 +432,7 @@ AVFilter ff_af_sidechaingate = {
.priv_size = sizeof(AudioGateContext),
.priv_class = &sidechaingate_class,
.query_formats = scquery_formats,
.uninit = uninit,
.inputs = sidechaingate_inputs,
.outputs = sidechaingate_outputs,
};
......
libavfilter/af_sidechaincompress.c
View file @ d4ce63a1
......@@ -24,6 +24,7 @@
* Audio (Sidechain) Compressor filter
*/
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
......@@ -57,7 +58,8 @@ typedef struct SidechainCompressContext {
int link;
int detection;
AVFrame *input_frame[2];
AVAudioFifo *fifo[2];
int64_t pts;
} SidechainCompressContext;
#define OFFSET(x) offsetof(SidechainCompressContext, x)
......@@ -186,53 +188,62 @@ static int filter_frame(AVFilterLink *link, AVFrame *frame)
AVFilterContext *ctx = link->dst;
SidechainCompressContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
const double *scsrc;
double *sample;
AVFrame *out, *in[2] = { NULL };
double *dst;
int nb_samples;
int ret, i;
int i;
for (i = 0; i < 2; i++)
if (link == ctx->inputs[i])
break;
av_assert0(i < 2 && !s->input_frame[i]);
s->input_frame[i] = frame;
av_assert0(i < 2);
av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
frame->nb_samples);
av_frame_free(&frame);
if (!s->input_frame[0] || !s->input_frame[1])
nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
if (!nb_samples)
return 0;
nb_samples = FFMIN(s->input_frame[0]->nb_samples,
s->input_frame[1]->nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);
for (i = 0; i < 2; i++) {
in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
if (!in[i]) {
av_frame_free(&in[0]);
av_frame_free(&in[1]);
return AVERROR(ENOMEM);
}
av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
}
sample = (double *)s->input_frame[0]->data[0];
scsrc = (const double *)s->input_frame[1]->data[0];
dst = (double *)out->data[0];
out->pts = s->pts;
s->pts += nb_samples;
compressor(s, sample, sample, scsrc, nb_samples,
compressor(s, (double *)in[0]->data[0], dst,
(double *)in[1]->data[0], nb_samples,
s->level_in, s->level_sc,
ctx->inputs[0], ctx->inputs[1]);
ret = ff_filter_frame(outlink, s->input_frame[0]);
s->input_frame[0] = NULL;
av_frame_free(&s->input_frame[1]);
av_frame_free(&in[0]);
av_frame_free(&in[1]);
return ret;
return ff_filter_frame(outlink, out);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SidechainCompressContext *s = ctx->priv;
int i, ret;
int i;
/* get a frame on each input */
for (i = 0; i < 2; i++) {
AVFilterLink *inlink = ctx->inputs[i];
if (!s->input_frame[i] &&
(ret = ff_request_frame(inlink)) < 0)
return ret;
/* request the same number of samples on all inputs */
if (i == 0)
ctx->inputs[1]->request_samples = s->input_frame[0]->nb_samples;
if (!av_audio_fifo_size(s->fifo[i]))
return ff_request_frame(inlink);
}
return 0;
......@@ -276,6 +287,7 @@ static int query_formats(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SidechainCompressContext *s = ctx->priv;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
......@@ -290,23 +302,33 @@ static int config_output(AVFilterLink *outlink)
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
if (!s->fifo[0] || !s->fifo[1])
return AVERROR(ENOMEM);
compressor_config_output(outlink);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
SidechainCompressContext *s = ctx->priv;
av_audio_fifo_free(s->fifo[0]);
av_audio_fifo_free(s->fifo[1]);
}
static const AVFilterPad sidechaincompress_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.needs_writable = 1,
.needs_fifo = 1,
},{
.name = "sidechain",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.needs_fifo = 1,
},
{ NULL }
};
......@@ -327,6 +349,7 @@ AVFilter ff_af_sidechaincompress = {
.priv_size = sizeof(SidechainCompressContext),
.priv_class = &sidechaincompress_class,
.query_formats = query_formats,
.uninit = uninit,
.inputs = sidechaincompress_inputs,
.outputs = sidechaincompress_outputs,
};
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment