[FFmpeg-cvslog] avfilter/af_sidechaincompress & af_agate: use audio fifo from lavu
Paul B Mahol
git at videolan.org
Fri Jan 15 21:51:16 CET 2016
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Jan 15 21:34:05 2016 +0100| [d4ce63a1bf2520be7015df78dd8b042abe456c23] | committer: Paul B Mahol
avfilter/af_sidechaincompress & af_agate: use audio fifo from lavu
Fixes regression causing segfault.
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d4ce63a1bf2520be7015df78dd8b042abe456c23
---
libavfilter/af_agate.c | 80 +++++++++++++++++++++++-------------
libavfilter/af_sidechaincompress.c | 77 ++++++++++++++++++++++------------
2 files changed, 102 insertions(+), 55 deletions(-)
diff --git a/libavfilter/af_agate.c b/libavfilter/af_agate.c
index 291e803..17e770f 100644
--- a/libavfilter/af_agate.c
+++ b/libavfilter/af_agate.c
@@ -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,
};
diff --git a/libavfilter/af_sidechaincompress.c b/libavfilter/af_sidechaincompress.c
index dac4605..a151a02 100644
--- a/libavfilter/af_sidechaincompress.c
+++ b/libavfilter/af_sidechaincompress.c
@@ -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,
};
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