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authorPaul B Mahol <onemda@gmail.com>2021-10-05 15:45:43 +0300
committerPaul B Mahol <onemda@gmail.com>2021-10-05 15:49:44 +0300
commit3fe49b51fcfbd36805eb8c2fc25a4bdc1943fbb2 (patch)
tree6cbc8eb104576322493b7e6cbf9eb7cab822cccf /libavfilter/af_speechnorm.c
parent2ee4077248dcd96e50efc0738e60519ffb9a4c7f (diff)
avfilter/af_speechnorm: fix possible memleak on error to make frame writable
Fix this by adding support for non-writable frames.
Diffstat (limited to 'libavfilter/af_speechnorm.c')
-rw-r--r--libavfilter/af_speechnorm.c38
1 files changed, 25 insertions, 13 deletions
diff --git a/libavfilter/af_speechnorm.c b/libavfilter/af_speechnorm.c
index f56ca8e558..56032932b2 100644
--- a/libavfilter/af_speechnorm.c
+++ b/libavfilter/af_speechnorm.c
@@ -84,7 +84,7 @@ typedef struct SpeechNormalizerContext {
void (*analyze_channel)(AVFilterContext *ctx, ChannelContext *cc,
const uint8_t *srcp, int nb_samples);
void (*filter_channels[2])(AVFilterContext *ctx,
- AVFrame *in, int nb_samples);
+ AVFrame *in, AVFrame *out, int nb_samples);
} SpeechNormalizerContext;
#define OFFSET(x) offsetof(SpeechNormalizerContext, x)
@@ -295,14 +295,15 @@ ANALYZE_CHANNEL(flt, float, 0.f, (float)MIN_PEAK)
#define FILTER_CHANNELS(name, ptype) \
static void filter_channels_## name (AVFilterContext *ctx, \
- AVFrame *in, int nb_samples) \
+ AVFrame *in, AVFrame *out, int nb_samples) \
{ \
SpeechNormalizerContext *s = ctx->priv; \
AVFilterLink *inlink = ctx->inputs[0]; \
\
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
- ptype *dst = (ptype *)in->extended_data[ch]; \
+ const ptype *src = (const ptype *)in->extended_data[ch]; \
+ ptype *dst = (ptype *)out->extended_data[ch]; \
const int bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); \
int n = 0; \
\
@@ -316,7 +317,7 @@ static void filter_channels_## name (AVFilterContext *ctx,
gain = cc->gain_state; \
consume_pi(cc, size); \
for (int i = n; !ctx->is_disabled && i < n + size; i++) \
- dst[i] *= gain; \
+ dst[i] = src[i] * gain; \
n += size; \
} \
} \
@@ -337,7 +338,8 @@ static float flerp(float min, float max, float mix)
#define FILTER_LINK_CHANNELS(name, ptype, tlerp) \
static void filter_link_channels_## name (AVFilterContext *ctx, \
- AVFrame *in, int nb_samples) \
+ AVFrame *in, AVFrame *out, \
+ int nb_samples) \
{ \
SpeechNormalizerContext *s = ctx->priv; \
AVFilterLink *inlink = ctx->inputs[0]; \
@@ -369,7 +371,8 @@ static void filter_link_channels_## name (AVFilterContext *ctx,
\
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
- ptype *dst = (ptype *)in->extended_data[ch]; \
+ const ptype *src = (const ptype *)in->extended_data[ch]; \
+ ptype *dst = (ptype *)out->extended_data[ch]; \
\
consume_pi(cc, min_size); \
if (cc->bypass) \
@@ -377,7 +380,7 @@ static void filter_link_channels_## name (AVFilterContext *ctx,
\
for (int i = n; !ctx->is_disabled && i < n + min_size; i++) { \
ptype g = tlerp(s->prev_gain, gain, (i - n) / (ptype)min_size); \
- dst[i] *= g; \
+ dst[i] = src[i] * g; \
} \
} \
\
@@ -398,7 +401,7 @@ static int filter_frame(AVFilterContext *ctx)
while (s->queue.available > 0) {
int min_pi_nb_samples;
- AVFrame *in;
+ AVFrame *in, *out;
in = ff_bufqueue_peek(&s->queue, 0);
if (!in)
@@ -410,16 +413,25 @@ static int filter_frame(AVFilterContext *ctx)
in = ff_bufqueue_get(&s->queue);
- ret = av_frame_make_writable(in);
- if (ret < 0)
- return ret;
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
- s->filter_channels[s->link](ctx, in, in->nb_samples);
+ s->filter_channels[s->link](ctx, in, out, in->nb_samples);
s->pts = in->pts + av_rescale_q(in->nb_samples, av_make_q(1, outlink->sample_rate),
outlink->time_base);
- return ff_filter_frame(outlink, in);
+ if (out != in)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
}
for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) {