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Diffstat (limited to 'libavfilter/af_amix.c')
-rw-r--r--libavfilter/af_amix.c241
1 files changed, 119 insertions, 122 deletions
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
index bfba1504ea..e18743e0b8 100644
--- a/libavfilter/af_amix.c
+++ b/libavfilter/af_amix.c
@@ -2,20 +2,20 @@
* Audio Mix Filter
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -44,9 +44,8 @@
#include "formats.h"
#include "internal.h"
-#define INPUT_OFF 0 /**< input has reached EOF */
#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_SHORTEST 1
@@ -110,7 +109,7 @@ static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
int samples = nb_samples;
while (samples > 0) {
FrameInfo *info = frame_list->list;
- av_assert0(info != NULL);
+ av_assert0(info);
if (info->nb_samples <= samples) {
samples -= info->nb_samples;
frame_list->list = info->next;
@@ -142,7 +141,7 @@ static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t p
frame_list->list = info;
frame_list->end = info;
} else {
- av_assert0(frame_list->end != NULL);
+ av_assert0(frame_list->end);
frame_list->end->next = info;
frame_list->end = info;
}
@@ -155,7 +154,7 @@ static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t p
typedef struct MixContext {
const AVClass *class; /**< class for AVOptions */
- AVFloatDSPContext fdsp;
+ AVFloatDSPContext *fdsp;
int nb_inputs; /**< number of inputs */
int active_inputs; /**< number of input currently active */
@@ -175,27 +174,22 @@ typedef struct MixContext {
#define OFFSET(x) offsetof(MixContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
-static const AVOption options[] = {
+#define F AV_OPT_FLAG_FILTERING_PARAM
+static const AVOption amix_options[] = {
{ "inputs", "Number of inputs.",
- OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A },
+ OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A|F },
{ "duration", "How to determine the end-of-stream.",
- OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A, "duration" },
- { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A, "duration" },
- { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" },
- { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A, "duration" },
+ 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" },
{ "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 },
- { NULL },
-};
-
-static const AVClass amix_class = {
- .class_name = "amix filter",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
+ OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
+ { NULL }
};
+AVFILTER_DEFINE_CLASS(amix);
/**
* Update the scaling factors to apply to each input during mixing.
@@ -214,7 +208,7 @@ static void calculate_scales(MixContext *s, int nb_samples)
}
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;
else
s->input_scale[i] = 0.0f;
@@ -237,11 +231,11 @@ static int config_output(AVFilterLink *outlink)
if (!s->frame_list)
return AVERROR(ENOMEM);
- s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));
+ s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
if (!s->fifos)
return AVERROR(ENOMEM);
- s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+ s->nb_channels = outlink->channels;
for (i = 0; i < s->nb_inputs; i++) {
s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
if (!s->fifos[i])
@@ -254,7 +248,7 @@ static int config_output(AVFilterLink *outlink)
memset(s->input_state, INPUT_ON, s->nb_inputs);
s->active_inputs = s->nb_inputs;
- s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));
+ s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
if (!s->input_scale)
return AVERROR(ENOMEM);
s->scale_norm = s->active_inputs;
@@ -269,18 +263,58 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
+static int calc_active_inputs(MixContext *s);
+
/**
* 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;
MixContext *s = ctx->priv;
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);
+ if (nb_samples == 0)
+ return 0;
+
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
@@ -292,7 +326,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
}
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;
av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
@@ -303,7 +337,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
plane_size = FFALIGN(plane_size, 16);
for (p = 0; p < planes; p++) {
- s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],
+ s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
(float *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
@@ -319,29 +353,6 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
}
/**
- * 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.
*/
static int request_samples(AVFilterContext *ctx, int min_samples)
@@ -353,19 +364,21 @@ static int request_samples(AVFilterContext *ctx, int min_samples)
for (i = 1; i < s->nb_inputs; i++) {
ret = 0;
- if (s->input_state[i] == INPUT_OFF)
+ if (!(s->input_state[i] & INPUT_ON))
continue;
- while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)
- ret = ff_request_frame(ctx->inputs[i]);
+ 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] = INPUT_OFF;
+ s->input_state[i] = 0;
continue;
}
} else if (ret < 0)
return ret;
}
- return 0;
+ return output_frame(ctx->outputs[0]);
}
/**
@@ -379,11 +392,11 @@ static int calc_active_inputs(MixContext *s)
int i;
int active_inputs = 0;
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;
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))
return AVERROR_EOF;
return 0;
@@ -394,66 +407,30 @@ static int request_frame(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
int ret;
- int wanted_samples, available_samples;
+ int wanted_samples;
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
- if (s->input_state[0] == INPUT_OFF) {
- ret = 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->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] = INPUT_OFF;
+ s->input_state[0] = 0;
if (s->nb_inputs == 1)
return AVERROR_EOF;
- else
- return AVERROR(EAGAIN);
- } else if (ret < 0)
- return ret;
+ return output_frame(ctx->outputs[0]);
+ }
+ return ret;
}
av_assert0(s->frame_list->nb_frames > 0);
wanted_samples = frame_list_next_frame_size(s->frame_list);
- if (s->active_inputs > 1) {
- 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);
+ return request_samples(ctx, wanted_samples);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
@@ -483,6 +460,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
buf->nb_samples);
+ av_frame_free(&buf);
+ return output_frame(outlink);
+
fail:
av_frame_free(&buf);
@@ -501,12 +481,16 @@ static av_cold int init(AVFilterContext *ctx)
snprintf(name, sizeof(name), "input%d", i);
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_strdup(name);
+ if (!pad.name)
+ return AVERROR(ENOMEM);
pad.filter_frame = filter_frame;
ff_insert_inpad(ctx, i, &pad);
}
- avpriv_float_dsp_init(&s->fdsp, 0);
+ s->fdsp = avpriv_float_dsp_alloc(0);
+ if (!s->fdsp)
+ return AVERROR(ENOMEM);
return 0;
}
@@ -525,6 +509,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->frame_list);
av_freep(&s->input_state);
av_freep(&s->input_scale);
+ av_freep(&s->fdsp);
for (i = 0; i < ctx->nb_inputs; i++)
av_freep(&ctx->input_pads[i].name);
@@ -533,12 +518,27 @@ static av_cold void uninit(AVFilterContext *ctx)
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
- ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
- ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);
- ff_set_common_formats(ctx, formats);
- ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());
- ff_set_common_samplerates(ctx, ff_all_samplerates());
+ AVFilterChannelLayouts *layouts;
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
+ (ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
+ (ret = ff_set_common_formats (ctx, formats)) < 0 ||
+ (ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
+ (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
+ goto fail;
return 0;
+fail:
+ if (layouts)
+ av_freep(&layouts->channel_layouts);
+ av_freep(&layouts);
+ return ret;
}
static const AVFilterPad avfilter_af_amix_outputs[] = {
@@ -552,17 +552,14 @@ static const AVFilterPad avfilter_af_amix_outputs[] = {
};
AVFilter ff_af_amix = {
- .name = "amix",
- .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
- .priv_size = sizeof(MixContext),
- .priv_class = &amix_class,
-
+ .name = "amix",
+ .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
+ .priv_size = sizeof(MixContext),
+ .priv_class = &amix_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
-
- .inputs = NULL,
- .outputs = avfilter_af_amix_outputs,
-
- .flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
+ .inputs = NULL,
+ .outputs = avfilter_af_amix_outputs,
+ .flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-13 13:18:42 +0000