summaryrefslogtreecommitdiff
path: root/libavfilter/af_amix.c
diff options
context:
space:
mode:
Diffstat (limited to 'libavfilter/af_amix.c')
-rw-r--r--libavfilter/af_amix.c88
1 files changed, 48 insertions, 40 deletions
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
index bfba1504ea..9a3cbd4dcd 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
*/
@@ -110,7 +110,7 @@ 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 +142,7 @@ 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 +155,7 @@ 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 +175,22 @@ 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.
@@ -237,7 +232,7 @@ 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);
@@ -254,7 +249,7 @@ 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;
@@ -303,7 +298,7 @@ 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);
}
@@ -501,12 +496,16 @@ 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 +524,7 @@ 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 +533,23 @@ static av_cold void uninit(AVFilterContext *ctx)
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts;
+ int ret;
+
+ layouts = ff_all_channel_layouts();
+
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
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());
- return 0;
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+ return ff_set_common_samplerates(ctx, ff_all_samplerates());
}
static const AVFilterPad avfilter_af_amix_outputs[] = {
@@ -552,17 +563,14 @@ 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,
};
View Lorry Controller Status