#include "audio.h"
#include "avfilter.h"
+#include "filters.h"
#include "formats.h"
#include "internal.h"
return 0;
}
+/* FIXME: use directly links fifo */
typedef struct MixContext {
const AVClass *class; /**< class for AVOptions */
OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
{ "duration", "How to determine the end-of-stream.",
OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
- { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A|F, "duration" },
- { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" },
- { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A|F, "duration" },
+ { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
+ { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
+ { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
{ "dropout_transition", "Transition time, in seconds, for volume "
"renormalization when an input stream ends.",
OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
return 0;
}
-static int calc_active_inputs(MixContext *s);
-
/**
* Read samples from the input FIFOs, mix, and write to the output link.
*/
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
AVFrame *out_buf, *in_buf;
- int nb_samples, ns, ret, i;
-
- ret = calc_active_inputs(s);
- if (ret < 0)
- return ret;
+ int nb_samples, ns, i;
if (s->input_state[0] & INPUT_ON) {
/* first input live: use the corresponding frame size */
nb_samples = FFMIN(nb_samples, ns);
}
}
- if (nb_samples == INT_MAX)
- return AVERROR_EOF;
+ if (nb_samples == INT_MAX) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
+ return 0;
+ }
}
s->next_pts = frame_list_next_pts(s->frame_list);
static int request_samples(AVFilterContext *ctx, int min_samples)
{
MixContext *s = ctx->priv;
- int i, ret;
+ int i;
av_assert0(s->nb_inputs > 1);
for (i = 1; i < s->nb_inputs; i++) {
- ret = 0;
- if (!(s->input_state[i] & INPUT_ON))
+ if (!(s->input_state[i] & INPUT_ON) ||
+ (s->input_state[i] & INPUT_EOF))
continue;
if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
continue;
- ret = ff_request_frame(ctx->inputs[i]);
- if (ret == AVERROR_EOF) {
- s->input_state[i] |= INPUT_EOF;
- if (av_audio_fifo_size(s->fifos[i]) == 0) {
- s->input_state[i] = 0;
- continue;
- }
- } else if (ret < 0)
- return ret;
+ ff_inlink_request_frame(ctx->inputs[i]);
}
return output_frame(ctx->outputs[0]);
}
return 0;
}
-static int request_frame(AVFilterLink *outlink)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = outlink->src;
- MixContext *s = ctx->priv;
- int ret;
- int wanted_samples;
-
- ret = calc_active_inputs(s);
- if (ret < 0)
- return ret;
-
- if (!(s->input_state[0] & INPUT_ON))
- return request_samples(ctx, 1);
-
- if (s->frame_list->nb_frames == 0) {
- ret = ff_request_frame(ctx->inputs[0]);
- if (ret == AVERROR_EOF) {
- s->input_state[0] = 0;
- if (s->nb_inputs == 1)
- return AVERROR_EOF;
- return output_frame(ctx->outputs[0]);
- }
- return ret;
- }
- av_assert0(s->frame_list->nb_frames > 0);
+ AVFilterLink *outlink = ctx->outputs[0];
+ MixContext *s = ctx->priv;
+ AVFrame *buf = NULL;
+ int i, ret;
- wanted_samples = frame_list_next_frame_size(s->frame_list);
+ for (i = 0; i < s->nb_inputs; i++) {
+ AVFilterLink *inlink = ctx->inputs[i];
+
+ if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
+ if (i == 0) {
+ int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
+ outlink->time_base);
+ ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
+ if (ret < 0) {
+ av_frame_free(&buf);
+ return ret;
+ }
+ }
- return request_samples(ctx, wanted_samples);
-}
+ ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+ buf->nb_samples);
+ if (ret < 0) {
+ av_frame_free(&buf);
+ return ret;
+ }
-static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
-{
- AVFilterContext *ctx = inlink->dst;
- MixContext *s = ctx->priv;
- AVFilterLink *outlink = ctx->outputs[0];
- int i, ret = 0;
+ av_frame_free(&buf);
- for (i = 0; i < ctx->nb_inputs; i++)
- if (ctx->inputs[i] == inlink)
- break;
- if (i >= ctx->nb_inputs) {
- av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
- ret = AVERROR(EINVAL);
- goto fail;
+ ret = output_frame(outlink);
+ if (ret < 0)
+ return ret;
+ }
}
- if (i == 0) {
- int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
- outlink->time_base);
- ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
- if (ret < 0)
- goto fail;
+ for (i = 0; i < s->nb_inputs; i++) {
+ int64_t pts;
+ int status;
+
+ if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ if (i == 0) {
+ s->input_state[i] = 0;
+ if (s->nb_inputs == 1) {
+ ff_outlink_set_status(outlink, status, pts);
+ return 0;
+ }
+ } else {
+ s->input_state[i] |= INPUT_EOF;
+ if (av_audio_fifo_size(s->fifos[i]) == 0) {
+ s->input_state[i] = 0;
+ }
+ }
+ }
+ }
}
- ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
- buf->nb_samples);
+ if (calc_active_inputs(s)) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
+ return 0;
+ }
- av_frame_free(&buf);
- return output_frame(outlink);
+ if (ff_outlink_frame_wanted(outlink)) {
+ int wanted_samples;
-fail:
- av_frame_free(&buf);
+ if (!(s->input_state[0] & INPUT_ON))
+ return request_samples(ctx, 1);
- return ret;
+ if (s->frame_list->nb_frames == 0) {
+ ff_inlink_request_frame(ctx->inputs[0]);
+ return 0;
+ }
+ av_assert0(s->frame_list->nb_frames > 0);
+
+ wanted_samples = frame_list_next_frame_size(s->frame_list);
+
+ return request_samples(ctx, wanted_samples);
+ }
+
+ return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
MixContext *s = ctx->priv;
- int i;
+ int i, ret;
for (i = 0; i < s->nb_inputs; i++) {
char name[32];
pad.name = av_strdup(name);
if (!pad.name)
return AVERROR(ENOMEM);
- pad.filter_frame = filter_frame;
- ff_insert_inpad(ctx, i, &pad);
+ if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
+ av_freep(&pad.name);
+ return ret;
+ }
}
s->fdsp = avpriv_float_dsp_alloc(0);
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
- .request_frame = request_frame
},
{ NULL }
};
.priv_class = &amix_class,
.init = init,
.uninit = uninit,
+ .activate = activate,
.query_formats = query_formats,
.inputs = NULL,
.outputs = avfilter_af_amix_outputs,
projects / ffmpeg / blobdiff