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avfilter/af_speechnorm: fix possible memleak on error to make frame writable 

Fix this by adding support for non-writable frames.
This commit is contained in:
Paul B Mahol 2021-10-05 14:45:43 +02:00
parent 2ee4077248
commit 3fe49b51fc
1 changed files with 25 additions and 13 deletions
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38
libavfilter/af_speechnorm.c
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@ -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++) {