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lavfi/af_amix: mostly fix scheduling.

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This commit is contained in:
Nicolas George
2015-10-24 15:19:11 +02:00
parent f53c4b6a18
commit a08fb3983f
1 changed files with 62 additions and 80 deletions
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142
libavfilter/af_amix.c
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@@ -44,9 +44,8 @@
#include "formats.h" #include "formats.h"
#include "internal.h" #include "internal.h"
#define INPUT_OFF 0 /**< input has reached EOF */
#define INPUT_ON 1 /**< input is active */ #define INPUT_ON 1 /**< input is active */
#define INPUT_INACTIVE 2 /**< input is on, but is currently inactive */ #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
#define DURATION_LONGEST 0 #define DURATION_LONGEST 0
#define DURATION_SHORTEST 1 #define DURATION_SHORTEST 1
@@ -209,7 +208,7 @@ static void calculate_scales(MixContext *s, int nb_samples)
} }
for (i = 0; i < s->nb_inputs; i++) { for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] == INPUT_ON) if (s->input_state[i] & INPUT_ON)
s->input_scale[i] = 1.0f / s->scale_norm; s->input_scale[i] = 1.0f / s->scale_norm;
else else
s->input_scale[i] = 0.0f; s->input_scale[i] = 0.0f;
@@ -264,15 +263,52 @@ static int config_output(AVFilterLink *outlink)
return 0; return 0;
} }
static int calc_active_inputs(MixContext *s);
/** /**
* Read samples from the input FIFOs, mix, and write to the output link. * Read samples from the input FIFOs, mix, and write to the output link.
*/ */
static int output_frame(AVFilterLink *outlink, int nb_samples) static int output_frame(AVFilterLink *outlink)
{ {
AVFilterContext *ctx = outlink->src; AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv; MixContext *s = ctx->priv;
AVFrame *out_buf, *in_buf; AVFrame *out_buf, *in_buf;
int i; int nb_samples, ns, ret, i;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
if (s->input_state[0] & INPUT_ON) {
/* first input live: use the corresponding frame size */
nb_samples = frame_list_next_frame_size(s->frame_list);
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
if (ns < nb_samples) {
if (!(s->input_state[i] & INPUT_EOF))
/* unclosed input with not enough samples */
return 0;
/* closed input to drain */
nb_samples = ns;
}
}
}
} else {
/* first input closed: use the available samples */
nb_samples = INT_MAX;
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
nb_samples = FFMIN(nb_samples, ns);
}
}
if (nb_samples == INT_MAX)
return AVERROR_EOF;
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, nb_samples);
calculate_scales(s, nb_samples); calculate_scales(s, nb_samples);
@@ -287,7 +323,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
} }
for (i = 0; i < s->nb_inputs; i++) { for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] == INPUT_ON) { if (s->input_state[i] & INPUT_ON) {
int planes, plane_size, p; int planes, plane_size, p;
av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data, av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
@@ -313,29 +349,6 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
return ff_filter_frame(outlink, out_buf); return ff_filter_frame(outlink, out_buf);
} }
/**
* Returns the smallest number of samples available in the input FIFOs other
* than that of the first input.
*/
static int get_available_samples(MixContext *s)
{
int i;
int available_samples = INT_MAX;
av_assert0(s->nb_inputs > 1);
for (i = 1; i < s->nb_inputs; i++) {
int nb_samples;
if (s->input_state[i] == INPUT_OFF)
continue;
nb_samples = av_audio_fifo_size(s->fifos[i]);
available_samples = FFMIN(available_samples, nb_samples);
}
if (available_samples == INT_MAX)
return 0;
return available_samples;
}
/** /**
* Requests a frame, if needed, from each input link other than the first. * Requests a frame, if needed, from each input link other than the first.
*/ */
@@ -348,19 +361,21 @@ static int request_samples(AVFilterContext *ctx, int min_samples)
for (i = 1; i < s->nb_inputs; i++) { for (i = 1; i < s->nb_inputs; i++) {
ret = 0; ret = 0;
if (s->input_state[i] == INPUT_OFF) if (!(s->input_state[i] & INPUT_ON))
continue; continue;
while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples) if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
ret = ff_request_frame(ctx->inputs[i]); continue;
ret = ff_request_frame(ctx->inputs[i]);
if (ret == AVERROR_EOF) { if (ret == AVERROR_EOF) {
s->input_state[i] |= INPUT_EOF;
if (av_audio_fifo_size(s->fifos[i]) == 0) { if (av_audio_fifo_size(s->fifos[i]) == 0) {
s->input_state[i] = INPUT_OFF; s->input_state[i] = 0;
continue; continue;
} }
} else if (ret < 0) } else if (ret < 0)
return ret; return ret;
} }
return 0; return output_frame(ctx->outputs[0]);
} }
/** /**
@@ -374,11 +389,11 @@ static int calc_active_inputs(MixContext *s)
int i; int i;
int active_inputs = 0; int active_inputs = 0;
for (i = 0; i < s->nb_inputs; i++) for (i = 0; i < s->nb_inputs; i++)
active_inputs += !!(s->input_state[i] != INPUT_OFF); active_inputs += !!(s->input_state[i] & INPUT_ON);
s->active_inputs = active_inputs; s->active_inputs = active_inputs;
if (!active_inputs || if (!active_inputs ||
(s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) || (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
(s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs)) (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
return AVERROR_EOF; return AVERROR_EOF;
return 0; return 0;
@@ -389,66 +404,30 @@ static int request_frame(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src; AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv; MixContext *s = ctx->priv;
int ret; int ret;
int wanted_samples, available_samples; int wanted_samples;
ret = calc_active_inputs(s); ret = calc_active_inputs(s);
if (ret < 0) if (ret < 0)
return ret; return ret;
if (s->input_state[0] == INPUT_OFF) { if (!(s->input_state[0] & INPUT_ON))
ret = request_samples(ctx, 1); return request_samples(ctx, 1);
if (ret < 0)
return ret;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
available_samples = get_available_samples(s);
if (!available_samples)
return AVERROR(EAGAIN);
return output_frame(outlink, available_samples);
}
if (s->frame_list->nb_frames == 0) { if (s->frame_list->nb_frames == 0) {
ret = ff_request_frame(ctx->inputs[0]); ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) { if (ret == AVERROR_EOF) {
s->input_state[0] = INPUT_OFF; s->input_state[0] = 0;
if (s->nb_inputs == 1) if (s->nb_inputs == 1)
return AVERROR_EOF; return AVERROR_EOF;
else return output_frame(ctx->outputs[0]);
return AVERROR(EAGAIN); }
} else if (ret < 0) return ret;
return ret;
} }
av_assert0(s->frame_list->nb_frames > 0); av_assert0(s->frame_list->nb_frames > 0);
wanted_samples = frame_list_next_frame_size(s->frame_list); wanted_samples = frame_list_next_frame_size(s->frame_list);
if (s->active_inputs > 1) { return request_samples(ctx, wanted_samples);
ret = request_samples(ctx, wanted_samples);
if (ret < 0)
return ret;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
}
if (s->active_inputs > 1) {
available_samples = get_available_samples(s);
if (!available_samples)
return AVERROR(EAGAIN);
available_samples = FFMIN(available_samples, wanted_samples);
} else {
available_samples = wanted_samples;
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, available_samples);
return output_frame(outlink, available_samples);
} }
static int filter_frame(AVFilterLink *inlink, AVFrame *buf) static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
@@ -478,6 +457,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
buf->nb_samples); buf->nb_samples);
av_frame_free(&buf);
return output_frame(outlink);
fail: fail:
av_frame_free(&buf); av_frame_free(&buf);