falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
Browse Source

avfilter/af_afade: acrossfade: switch to activate 

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n4.0
Paul B Mahol 7 years ago
parent
1b5d3c08e3
commit
8088b5d69c
1 changed files with 83 additions and 124 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +83
    -124
      libavfilter/af_afade.c

+ 83
- 124
libavfilter/af_afade.c View File

@@ -23,10 +23,14 @@
* fade audio filter
*/

#define FF_INTERNAL_FIELDS 1
#include "framequeue.h"

#include "libavutil/audio_fifo.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"

typedef struct AudioFadeContext {
@@ -39,6 +43,7 @@ typedef struct AudioFadeContext {
int64_t start_time;
int overlap;
int cf0_eof;
int prev_size;
int crossfade_is_over;
AVAudioFifo *fifo[2];
int64_t pts;
@@ -428,157 +433,127 @@ CROSSFADE(flt, float)
CROSSFADE(s16, int16_t)
CROSSFADE(s32, int32_t)

static int acrossfade_filter_frame(AVFilterLink *inlink, AVFrame *in)
static int activate(AVFilterContext *ctx)
{
AVFilterContext *ctx = inlink->dst;
AudioFadeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out, *cf[2] = { NULL };
int ret = 0, nb_samples;
AVFrame *in = NULL, *out, *cf[2] = { NULL };
int ret = 0, nb_samples, status;
int64_t pts;

if (s->crossfade_is_over) {
ret = ff_inlink_consume_frame(ctx->inputs[1], &in);
if (ret < 0) {
return ret;
} else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
} else {
if (ff_outlink_frame_wanted(ctx->outputs[0]) && !in) {
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}
}
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
} else if (inlink == ctx->inputs[0]) {
av_audio_fifo_write(s->fifo[0], (void **)in->extended_data, in->nb_samples);
}

nb_samples = av_audio_fifo_size(s->fifo[0]) - s->nb_samples;
if (ff_framequeue_queued_samples(&ctx->inputs[0]->fifo) > s->nb_samples) {
nb_samples = ff_framequeue_queued_samples(&ctx->inputs[0]->fifo) - s->nb_samples;
if (nb_samples > 0) {
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
av_audio_fifo_read(s->fifo[0], (void **)out->extended_data, nb_samples);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
}
} else if (av_audio_fifo_size(s->fifo[1]) < s->nb_samples) {
if (!s->overlap && av_audio_fifo_size(s->fifo[0]) > 0) {
nb_samples = av_audio_fifo_size(s->fifo[0]);

cf[0] = ff_get_audio_buffer(outlink, nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out || !cf[0]) {
ret = AVERROR(ENOMEM);
goto fail;
ret = ff_inlink_consume_samples(ctx->inputs[0], nb_samples, nb_samples, &in);
if (ret < 0) {
return ret;
}
av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, nb_samples);

s->fade_samples(out->extended_data, cf[0]->extended_data, nb_samples,
outlink->channels, -1, nb_samples - 1, nb_samples, s->curve);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
}
av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
} else if (av_audio_fifo_size(s->fifo[1]) >= s->nb_samples) {
av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
in->pts = s->pts;
s->pts += av_rescale_q(in->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
return ff_filter_frame(outlink, in);
} else if (ff_framequeue_queued_samples(&ctx->inputs[1]->fifo) >= s->nb_samples) {
if (s->overlap) {
cf[0] = ff_get_audio_buffer(outlink, s->nb_samples);
cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out || !cf[0] || !cf[1]) {
if (!out)
return AVERROR(ENOMEM);

ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
ret = AVERROR(ENOMEM);
goto fail;
return ret;
}

av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, s->nb_samples);
av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}

s->crossfade_samples(out->extended_data, cf[0]->extended_data,
cf[1]->extended_data,
s->nb_samples, in->channels,
s->nb_samples, out->channels,
s->curve, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
s->crossfade_is_over = 1;
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
} else {
out = ff_get_audio_buffer(outlink, s->nb_samples);
cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out || !cf[1]) {
ret = AVERROR(ENOMEM);
if (!out)
return AVERROR(ENOMEM);

ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]);
if (ret < 0) {
av_frame_free(&out);
goto fail;
return ret;
}

av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);

s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
outlink->channels, 1, 0, s->nb_samples, s->curve2);
s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples,
outlink->channels, -1, s->nb_samples - 1, s->nb_samples, s->curve);
out->pts = s->pts;
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
av_frame_free(&cf[0]);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto fail;
}
return ret;

nb_samples = av_audio_fifo_size(s->fifo[1]);
if (nb_samples > 0) {
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
out = ff_get_audio_buffer(outlink, s->nb_samples);
if (!out)
return AVERROR(ENOMEM);

ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]);
if (ret < 0) {
av_frame_free(&out);
return ret;
}

av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
outlink->channels, 1, 0, s->nb_samples, s->curve2);
out->pts = s->pts;
s->pts += av_rescale_q(nb_samples,
s->pts += av_rescale_q(s->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
ret = ff_filter_frame(outlink, out);
s->crossfade_is_over = 1;
av_frame_free(&cf[1]);
return ff_filter_frame(outlink, out);
}
s->crossfade_is_over = 1;
}

fail:
av_frame_free(&in);
av_frame_free(&cf[0]);
av_frame_free(&cf[1]);
return ret;
}

static int acrossfade_request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFadeContext *s = ctx->priv;
int ret = 0;

if (!s->cf0_eof) {
AVFilterLink *cf0 = ctx->inputs[0];
ret = ff_request_frame(cf0);
if (ret < 0 && ret != AVERROR_EOF)
return ret;
if (ret == AVERROR_EOF) {
} else if (ff_outlink_frame_wanted(ctx->outputs[0])) {
if (!s->cf0_eof && ctx->inputs[0]->status_in) {
s->cf0_eof = 1;
ret = 0;
}
} else {
AVFilterLink *cf1 = ctx->inputs[1];
int nb_samples = av_audio_fifo_size(s->fifo[1]);

ret = ff_request_frame(cf1);
if (ret == AVERROR_EOF && nb_samples > 0) {
AVFrame *out = ff_get_audio_buffer(outlink, nb_samples);
if (!out)
return AVERROR(ENOMEM);

av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
ret = ff_filter_frame(outlink, out);
if (ctx->inputs[1]->status_in) {
ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, AV_NOPTS_VALUE);
return 0;
}
if (!s->cf0_eof)
ff_inlink_request_frame(ctx->inputs[0]);
else
ff_inlink_request_frame(ctx->inputs[1]);
return 0;
}

return ret;
@@ -615,32 +590,17 @@ static int acrossfade_config_output(AVFilterLink *outlink)

config_output(outlink);

s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
if (!s->fifo[0] || !s->fifo[1])
return AVERROR(ENOMEM);

return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
AudioFadeContext *s = ctx->priv;

av_audio_fifo_free(s->fifo[0]);
av_audio_fifo_free(s->fifo[1]);
}

static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
{
.name = "crossfade0",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = acrossfade_filter_frame,
},
{
.name = "crossfade1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = acrossfade_filter_frame,
},
{ NULL }
};
@@ -649,7 +609,6 @@ static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = acrossfade_request_frame,
.config_props = acrossfade_config_output,
},
{ NULL }
@@ -660,7 +619,7 @@ AVFilter ff_af_acrossfade = {
.description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
.query_formats = query_formats,
.priv_size = sizeof(AudioFadeContext),
.uninit = uninit,
.activate = activate,
.priv_class = &acrossfade_class,
.inputs = avfilter_af_acrossfade_inputs,
.outputs = avfilter_af_acrossfade_outputs,


Write Preview
Loading…
Cancel
Save