[FFmpeg-cvslog] avfilter/af_headphone: switch to activate
Paul B Mahol
git at videolan.org
Mon Apr 16 19:45:05 EEST 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Apr 16 16:08:55 2018 +0200| [9cf0079638e1e625305ba049cb3c1f57e88c1b49] | committer: Paul B Mahol
avfilter/af_headphone: switch to activate
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9cf0079638e1e625305ba049cb3c1f57e88c1b49
---
libavfilter/af_headphone.c | 154 ++++++++++++++++++++-------------------------
1 file changed, 67 insertions(+), 87 deletions(-)
diff --git a/libavfilter/af_headphone.c b/libavfilter/af_headphone.c
index 7910132218..6b210e1436 100644
--- a/libavfilter/af_headphone.c
+++ b/libavfilter/af_headphone.c
@@ -29,6 +29,7 @@
#include "libavcodec/avfft.h"
#include "avfilter.h"
+#include "filters.h"
#include "internal.h"
#include "audio.h"
@@ -48,7 +49,6 @@ typedef struct HeadphoneContext {
int have_hrirs;
int eof_hrirs;
- int64_t pts;
int ir_len;
@@ -328,15 +328,11 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
return 0;
}
-static int read_ir(AVFilterLink *inlink, AVFrame *frame)
+static int read_ir(AVFilterLink *inlink, int input_number, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
- int ir_len, max_ir_len, input_number, ret;
-
- for (input_number = 0; input_number < s->nb_inputs; input_number++)
- if (inlink == ctx->inputs[input_number])
- break;
+ int ir_len, max_ir_len, ret;
ret = av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data,
frame->nb_samples);
@@ -357,22 +353,19 @@ static int read_ir(AVFilterLink *inlink, AVFrame *frame)
return 0;
}
-static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink, int max_nb_samples)
+static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
- AVFrame *in = s->in[0].frame;
int n_clippings[2] = { 0 };
ThreadData td;
AVFrame *out;
- av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size);
-
out = ff_get_audio_buffer(outlink, in->nb_samples);
- if (!out)
+ if (!out) {
+ av_frame_free(&in);
return AVERROR(ENOMEM);
- out->pts = s->pts;
- if (s->pts != AV_NOPTS_VALUE)
- s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+ }
+ out->pts = in->pts;
td.in = in; td.out = out; td.write = s->write;
td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
@@ -391,7 +384,7 @@ static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink, int max_n
n_clippings[0] + n_clippings[1], out->nb_samples * 2);
}
- out->nb_samples = max_nb_samples;
+ av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
@@ -464,11 +457,6 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
goto fail;
}
- s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size);
- if (!s->in[0].frame) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
for (i = 0; i < s->nb_inputs - 1; i++) {
s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len);
if (!s->in[i + 1].frame) {
@@ -624,22 +612,58 @@ fail:
return ret;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
- int ret = 0;
+ AVFrame *in = NULL;
+ int i, ret;
- ret = av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
- in->nb_samples);
- if (s->pts == AV_NOPTS_VALUE)
- s->pts = in->pts;
+ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+ if (!s->eof_hrirs) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ AVFrame *ir = NULL;
+ int64_t pts;
+ int status;
- av_frame_free(&in);
+ if (s->in[i].eof)
+ continue;
- if (ret < 0)
- return ret;
+ if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &ir)) > 0) {
+ ret = read_ir(ctx->inputs[i], i, ir);
+ if (ret < 0)
+ return ret;
+ }
+ if (ret < 0)
+ return ret;
+
+ if (!s->in[i].eof) {
+ if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ s->in[i].eof = 1;
+ }
+ }
+ }
+ }
+
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ break;
+ }
+
+ if (i != s->nb_inputs) {
+ if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ ff_inlink_request_frame(ctx->inputs[i]);
+ }
+ }
+ return 0;
+ } else {
+ s->eof_hrirs = 1;
+ }
+ }
if (!s->have_hrirs && s->eof_hrirs) {
ret = convert_coeffs(ctx, inlink);
@@ -647,14 +671,19 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
return ret;
}
- if (s->have_hrirs) {
- while (av_audio_fifo_size(s->in[0].fifo) >= s->size) {
- ret = headphone_frame(s, outlink, s->size);
- if (ret < 0)
- return ret;
- }
+ if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
+ ret = headphone_frame(s, in, outlink);
+ if (ret < 0)
+ return ret;
}
+ if (ret < 0)
+ return ret;
+
+ FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
+ if (ff_outlink_frame_wanted(ctx->outputs[0]))
+ ff_inlink_request_frame(ctx->inputs[0]);
+
return 0;
}
@@ -733,7 +762,6 @@ static av_cold int init(AVFilterContext *ctx)
.name = "in0",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
- .filter_frame = filter_frame,
};
if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
return ret;
@@ -754,7 +782,6 @@ static av_cold int init(AVFilterContext *ctx)
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = read_ir,
};
if (!name)
return AVERROR(ENOMEM);
@@ -767,7 +794,6 @@ static av_cold int init(AVFilterContext *ctx)
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
- s->pts = AV_NOPTS_VALUE;
return 0;
}
@@ -798,52 +824,6 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
-static int request_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- HeadphoneContext *s = ctx->priv;
- int i, ret;
-
- for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) {
- if (!s->in[i].eof) {
- ret = ff_request_frame(ctx->inputs[i]);
- if (ret == AVERROR_EOF) {
- s->in[i].eof = 1;
- ret = 0;
- }
- return ret;
- } else {
- if (i == s->nb_inputs - 1)
- s->eof_hrirs = 1;
- }
- }
-
- ret = ff_request_frame(ctx->inputs[0]);
- if (ret == AVERROR_EOF && av_audio_fifo_size(s->in[0].fifo) > 0 && s->have_hrirs) {
- int nb_samples = av_audio_fifo_size(s->in[0].fifo);
- AVFrame *in = ff_get_audio_buffer(ctx->inputs[0], s->size - nb_samples);
-
- if (!in)
- return AVERROR(ENOMEM);
-
- av_samples_set_silence(in->extended_data, 0,
- in->nb_samples,
- in->channels,
- in->format);
-
- ret = av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
- in->nb_samples);
- av_frame_free(&in);
- if (ret < 0)
- return ret;
- ret = headphone_frame(s, outlink, nb_samples);
-
- av_audio_fifo_drain(s->in[0].fifo, av_audio_fifo_size(s->in[0].fifo));
- }
-
- return ret;
-}
-
static av_cold void uninit(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
@@ -900,7 +880,6 @@ static const AVFilterPad outputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
- .request_frame = request_frame,
},
{ NULL }
};
@@ -913,6 +892,7 @@ AVFilter ff_af_headphone = {
.init = init,
.uninit = uninit,
.query_formats = query_formats,
+ .activate = activate,
.inputs = NULL,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
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