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authorMichael Niedermayer <michaelni@gmx.at>2012-12-18 17:22:36 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-12-18 17:30:31 +0100
commitc311431e291ee90cddb074ef4ba18496755a7406 (patch)
treed7cdcec0e939ac3d0a03bf1b8643d8a6a069fe3f /libavfilter/af_apad.c
parentc2640146e06faa7608a6bfecc36fc66d6a920e6e (diff)
downloadffmpeg-c311431e291ee90cddb074ef4ba18496755a7406.tar.gz
apad: add whole_len and pad_len options.
These options allow padding to a specific length or adding a specific amount if silence. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_apad.c')
-rw-r--r--libavfilter/af_apad.c29
1 files changed, 28 insertions, 1 deletions
diff --git a/libavfilter/af_apad.c b/libavfilter/af_apad.c
index 823aa969fe..4bf2d8111b 100644
--- a/libavfilter/af_apad.c
+++ b/libavfilter/af_apad.c
@@ -40,6 +40,8 @@ typedef struct {
int64_t next_pts;
int packet_size;
+ int64_t pad_len;
+ int64_t whole_len;
} APadContext;
#define OFFSET(x) offsetof(APadContext, x)
@@ -47,6 +49,8 @@ typedef struct {
static const AVOption apad_options[] = {
{ "packet_size", "set silence packet size", OFFSET(packet_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 0, INT_MAX, A },
+ { "pad_len", "number of samples of silence to add", OFFSET(pad_len), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
+ { "whole_len", "target number of samples in the audio stream", OFFSET(whole_len), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
{ NULL },
};
@@ -65,6 +69,11 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
if ((ret = av_opt_set_from_string(apad, args, NULL, "=", ":")) < 0)
return ret;
+ if (apad->whole_len && apad->pad_len) {
+ av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
+ return AVERROR(EINVAL);
+ }
+
av_opt_free(apad);
return 0;
@@ -74,6 +83,10 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *frame)
{
AVFilterContext *ctx = inlink->dst;
APadContext *apad = ctx->priv;
+
+ if (apad->whole_len)
+ apad->whole_len -= frame->audio->nb_samples;
+
apad->next_pts = frame->pts + av_rescale_q(frame->audio->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
return ff_filter_frame(ctx->outputs[0], frame);
}
@@ -88,7 +101,21 @@ static int request_frame(AVFilterLink *outlink)
if (ret == AVERROR_EOF) {
int n_out = apad->packet_size;
- AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+ AVFilterBufferRef *outsamplesref;
+
+ if (apad->whole_len > 0) {
+ apad->pad_len = apad->whole_len;
+ apad->whole_len = 0;
+ }
+ if (apad->pad_len > 0) {
+ n_out = FFMIN(n_out, apad->pad_len);
+ apad->pad_len -= n_out;
+ }
+
+ if(!n_out)
+ return AVERROR_EOF;
+
+ outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
if (!outsamplesref)
return AVERROR(ENOMEM);