Browse Source

avfilter: pass outlink to ff_get_audio_buffer()

This is more correct.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n4.0
Paul B Mahol 8 years ago
parent
commit
88cbd25b19
27 changed files with 29 additions and 29 deletions
  1. +1
    -1
      libavfilter/af_acontrast.c
  2. +1
    -1
      libavfilter/af_adelay.c
  3. +1
    -1
      libavfilter/af_aecho.c
  4. +1
    -1
      libavfilter/af_aemphasis.c
  5. +1
    -1
      libavfilter/af_afade.c
  6. +1
    -1
      libavfilter/af_agate.c
  7. +1
    -1
      libavfilter/af_alimiter.c
  8. +1
    -1
      libavfilter/af_aphaser.c
  9. +1
    -1
      libavfilter/af_biquads.c
  10. +1
    -1
      libavfilter/af_bs2b.c
  11. +1
    -1
      libavfilter/af_chorus.c
  12. +2
    -2
      libavfilter/af_compand.c
  13. +1
    -1
      libavfilter/af_compensationdelay.c
  14. +1
    -1
      libavfilter/af_crossfeed.c
  15. +1
    -1
      libavfilter/af_earwax.c
  16. +1
    -1
      libavfilter/af_extrastereo.c
  17. +1
    -1
      libavfilter/af_flanger.c
  18. +1
    -1
      libavfilter/af_haas.c
  19. +1
    -1
      libavfilter/af_loudnorm.c
  20. +1
    -1
      libavfilter/af_replaygain.c
  21. +2
    -2
      libavfilter/af_rubberband.c
  22. +1
    -1
      libavfilter/af_sidechaincompress.c
  23. +1
    -1
      libavfilter/af_stereotools.c
  24. +1
    -1
      libavfilter/af_stereowiden.c
  25. +1
    -1
      libavfilter/af_tremolo.c
  26. +1
    -1
      libavfilter/af_vibrato.c
  27. +1
    -1
      libavfilter/af_volume.c

+ 1
- 1
libavfilter/af_acontrast.c View File

@@ -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);


+ 1
- 1
libavfilter/af_adelay.c View File

@@ -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);


+ 1
- 1
libavfilter/af_aecho.c View File

@@ -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);


+ 1
- 1
libavfilter/af_aemphasis.c View File

@@ -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);


+ 1
- 1
libavfilter/af_afade.c View File

@@ -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);


+ 1
- 1
libavfilter/af_agate.c View File

@@ -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);


+ 1
- 1
libavfilter/af_alimiter.c View File

@@ -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);


+ 1
- 1
libavfilter/af_aphaser.c View File

@@ -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);


+ 1
- 1
libavfilter/af_biquads.c View File

@@ -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);


+ 1
- 1
libavfilter/af_bs2b.c View File

@@ -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);


+ 1
- 1
libavfilter/af_chorus.c View File

@@ -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);


+ 2
- 2
libavfilter/af_compand.c View File

@@ -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);


+ 1
- 1
libavfilter/af_compensationdelay.c View File

@@ -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);


+ 1
- 1
libavfilter/af_crossfeed.c View File

@@ -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);


+ 1
- 1
libavfilter/af_earwax.c View File

@@ -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) {


+ 1
- 1
libavfilter/af_extrastereo.c View File

@@ -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);


+ 1
- 1
libavfilter/af_flanger.c View File

@@ -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);


+ 1
- 1
libavfilter/af_haas.c View File

@@ -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);


+ 1
- 1
libavfilter/af_loudnorm.c View File

@@ -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);


+ 1
- 1
libavfilter/af_replaygain.c View File

@@ -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);


+ 2
- 2
libavfilter/af_rubberband.c View File

@@ -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,


+ 1
- 1
libavfilter/af_sidechaincompress.c View File

@@ -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);


+ 1
- 1
libavfilter/af_stereotools.c View File

@@ -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);


+ 1
- 1
libavfilter/af_stereowiden.c View File

@@ -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);


+ 1
- 1
libavfilter/af_tremolo.c View File

@@ -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);


+ 1
- 1
libavfilter/af_vibrato.c View File

@@ -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);


+ 1
- 1
libavfilter/af_volume.c View File

@@ -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);


Loading…
Cancel
Save