avfilter: pass outlink to ff_get_audio_buffer()

This is more correct.
Signed-off-by: Paul B Mahol <onemda@gmail.com>

libavfilter/af_acontrast.c

@@ -173,7 +173,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_adelay.c

@@ -192,7 +192,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     if (ctx->is_disabled || !s->delays)
         return ff_filter_frame(ctx->outputs[0], frame);
 
-    out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+    out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
     if (!out_frame) {
         av_frame_free(&frame);
         return AVERROR(ENOMEM);

libavfilter/af_aecho.c

@@ -279,7 +279,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
-        out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+        out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);

libavfilter/af_aemphasis.c

@@ -96,7 +96,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_afade.c

@@ -282,7 +282,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
     if (av_frame_is_writable(buf)) {
         out_buf = buf;
     } else {
-        out_buf = ff_get_audio_buffer(inlink, nb_samples);
+        out_buf = ff_get_audio_buffer(outlink, nb_samples);
         if (!out_buf)
             return AVERROR(ENOMEM);
         av_frame_copy_props(out_buf, buf);

libavfilter/af_agate.c

@@ -214,7 +214,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_alimiter.c

@@ -135,7 +135,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_aphaser.c

@@ -247,7 +247,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
     if (av_frame_is_writable(inbuf)) {
         outbuf = inbuf;
     } else {
-        outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
+        outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
         if (!outbuf) {
             av_frame_free(&inbuf);
             return AVERROR(ENOMEM);

libavfilter/af_biquads.c

@@ -417,7 +417,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
     if (av_frame_is_writable(buf)) {
         out_buf = buf;
     } else {
-        out_buf = ff_get_audio_buffer(inlink, nb_samples);
+        out_buf = ff_get_audio_buffer(outlink, nb_samples);
         if (!out_buf) {
             av_frame_free(&buf);
             return AVERROR(ENOMEM);

libavfilter/af_bs2b.c

@@ -135,7 +135,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
-        out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+        out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);

libavfilter/af_chorus.c

@@ -247,7 +247,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
-        out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+        out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);

libavfilter/af_compand.c

@@ -185,7 +185,7 @@ static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
-        out_frame = ff_get_audio_buffer(inlink, nb_samples);
+        out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);
@@ -249,7 +249,7 @@ static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
 
             if (count >= s->delay_samples) {
                 if (!out_frame) {
-                    out_frame = ff_get_audio_buffer(inlink, nb_samples - i);
+                    out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples - i);
                     if (!out_frame) {
                         av_frame_free(&frame);
                         return AVERROR(ENOMEM);

libavfilter/af_compensationdelay.c

@@ -131,7 +131,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     AVFrame *out;
     int n, ch;
 
-    out = ff_get_audio_buffer(inlink, in->nb_samples);
+    out = ff_get_audio_buffer(ctx->outputs[0], in->nb_samples);
     if (!out) {
         av_frame_free(&in);
         return AVERROR(ENOMEM);

libavfilter/af_crossfeed.c

@@ -99,7 +99,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_earwax.c

@@ -115,7 +115,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
 {
     AVFilterLink *outlink = inlink->dst->outputs[0];
     int16_t *taps, *endin, *in, *out;
-    AVFrame *outsamples = ff_get_audio_buffer(inlink, insamples->nb_samples);
+    AVFrame *outsamples = ff_get_audio_buffer(outlink, insamples->nb_samples);
     int len;
 
     if (!outsamples) {

libavfilter/af_extrastereo.c

@@ -71,7 +71,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_flanger.c

@@ -148,7 +148,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
-        out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+        out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);

libavfilter/af_haas.c

@@ -144,7 +144,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_loudnorm.c

@@ -423,7 +423,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_replaygain.c

@@ -554,7 +554,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     uint32_t level;
     AVFrame *out;
 
-    out = ff_get_audio_buffer(inlink, in->nb_samples);
+    out = ff_get_audio_buffer(outlink, in->nb_samples);
     if (!out) {
         av_frame_free(&in);
         return AVERROR(ENOMEM);

libavfilter/af_rubberband.c

@@ -128,7 +128,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 
     nb_samples = rubberband_available(s->rbs);
     if (nb_samples > 0) {
-        out = ff_get_audio_buffer(inlink, nb_samples);
+        out = ff_get_audio_buffer(outlink, nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
@@ -187,7 +187,7 @@ static int request_frame(AVFilterLink *outlink)
             nb_samples = rubberband_available(s->rbs);
 
             if (nb_samples > 0) {
-                out = ff_get_audio_buffer(inlink, nb_samples);
+                out = ff_get_audio_buffer(outlink, nb_samples);
                 if (!out)
                     return AVERROR(ENOMEM);
                 out->pts = av_rescale_q(s->nb_samples_out,

libavfilter/af_sidechaincompress.c

@@ -367,7 +367,7 @@ static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_stereotools.c

@@ -166,7 +166,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_stereowiden.c

@@ -98,7 +98,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_tremolo.c

@@ -57,7 +57,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_vibrato.c

@@ -63,7 +63,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     if (av_frame_is_writable(in)) {
         out = in;
     } else {
-        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);

libavfilter/af_volume.c

@@ -410,7 +410,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
             && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
         out_buf = buf;
     } else {
-        out_buf = ff_get_audio_buffer(inlink, nb_samples);
+        out_buf = ff_get_audio_buffer(outlink, nb_samples);
         if (!out_buf) {
             av_frame_free(&buf);
             return AVERROR(ENOMEM);