diff options
Diffstat (limited to 'libavfilter/af_amix.c')
| -rw-r--r-- | libavfilter/af_amix.c | 65 |
1 files changed, 32 insertions, 33 deletions
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c index bfba150..47cbb45 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; } @@ -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. @@ -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; @@ -533,10 +528,17 @@ static av_cold void uninit(AVFilterContext *ctx) static int query_formats(AVFilterContext *ctx) { AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts; + + 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_channel_layouts(ctx, layouts); ff_set_common_samplerates(ctx, ff_all_samplerates()); return 0; } @@ -552,17 +554,14 @@ 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, }; |
