[FFmpeg-devel] [PATCH] Add 'asetframesize' audiofilter

Andrey Utkin andrey.krieger.utkin at gmail.com
Tue Mar 13 16:32:39 CET 2012


This patch is resubmit of the one already reviewed by S. Sabatini
http://ffmpeg.org/pipermail/ffmpeg-devel/2012-February/121381.html
I've fixed what i could.
I found the following issues pointless or disappointing:
1. Is there any reason for which you can't set a default value? (for framesize)
2. AVCODEC_MAX_AUDIO_FRAME_SIZE non-availability
3. Possible int overflow (at av_fifo_realloc2)

I request your attention and help to tricky moment of the algorithm: this
filter does not call avfilter_filter_samples() from its .filter_samples() proc;
but all other audiofilters do. Is that wrong, or ok?
---8<---
Filter that changes number of samples on single output operation
---
 libavfilter/Makefile           |    1 +
 libavfilter/af_asetframesize.c |  199 ++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c       |    1 +
 3 files changed, 201 insertions(+), 0 deletions(-)
 create mode 100644 libavfilter/af_asetframesize.c

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e9c9a4b..1938312 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -31,6 +31,7 @@ OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
+OBJS-$(CONFIG_ASETFRAMESIZE_FILTER)          += af_asetframesize.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
 OBJS-$(CONFIG_ASPLIT_FILTER)                 += af_asplit.o
 OBJS-$(CONFIG_ASTREAMSYNC_FILTER)            += af_astreamsync.o
diff --git a/libavfilter/af_asetframesize.c b/libavfilter/af_asetframesize.c
new file mode 100644
index 0000000..8f13893
--- /dev/null
+++ b/libavfilter/af_asetframesize.c
@@ -0,0 +1,199 @@
+/*
+ * Copyright (c) 2012 Andrey Utkin
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Filter that changes number of samples on single output operation
+ */
+
+#include "avfilter.h"
+#include "libavutil/fifo.h"
+
+typedef struct {
+    unsigned int nb_out_samples; ///< how many samples to output at once
+    AVFifoBuffer *fifo;          ///< samples are queued here
+    int64_t next_out_pts;
+} ASFSContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+    ASFSContext *asfs = ctx->priv;
+    int ret;
+    if (!args) {
+        av_log(ctx, AV_LOG_ERROR, "Parameter is required\n");
+        return AVERROR(EINVAL);
+    }
+    ret = sscanf(args, "%u", &asfs->nb_out_samples);
+    if ((ret != 1) || (!asfs->nb_out_samples)) {
+        av_log(ctx, AV_LOG_ERROR, "Invalid specified framesize '%s'\n", args);
+        return AVERROR(EINVAL);
+    }
+
+    asfs->fifo = av_fifo_alloc(2 * AVCODEC_MAX_AUDIO_FRAME_SIZE);
+    if (!asfs->fifo)
+        return AVERROR(ENOMEM);
+
+    asfs->next_out_pts = AV_NOPTS_VALUE;
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ASFSContext *asfs = ctx->priv;
+    av_fifo_free(asfs->fifo);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    /*
+     * State that we support:
+     * - every sample format
+     * - every channels layout
+     * - only PACKED packing format
+     */
+    AVFilterFormats *formats = NULL;
+    int packing_fmts[] = { AVFILTER_PACKED, -1 };
+
+    formats = avfilter_make_all_channel_layouts();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_channel_layouts(ctx, formats);
+
+    formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_sample_formats(ctx, formats);
+
+    formats = avfilter_make_format_list(packing_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    avfilter_set_common_packing_formats(ctx, formats);
+
+    return 0;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+    int ret;
+    ASFSContext *asfs = inlink->dst->priv;
+    int nb_samples = insamples->audio->nb_samples;
+    int nb_channels =
+        av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
+    int bytes_per_sample_1ch = av_get_bytes_per_sample(insamples->format);
+    int nb_bytes_in  = nb_channels * bytes_per_sample_1ch * nb_samples;
+
+    if (av_fifo_space(asfs->fifo) < nb_bytes_in) {
+        av_log(inlink->dst, AV_LOG_DEBUG, "No space for %d bytes, stretching fifo\n", nb_bytes_in);
+        ret = av_fifo_realloc2(asfs->fifo, av_fifo_size(asfs->fifo) + nb_bytes_in);
+        if (ret < 0) {
+            av_log(inlink->dst, AV_LOG_ERROR,
+                    "Stretching fifo fail, discarded %d samples\n", nb_samples);
+            return;
+        }
+    }
+    av_fifo_generic_write(asfs->fifo, insamples->data[0], nb_bytes_in, NULL);
+
+    if (asfs->next_out_pts == AV_NOPTS_VALUE)
+        asfs->next_out_pts = insamples->pts;
+    avfilter_unref_buffer(insamples);
+}
+
+static int poll_frame(AVFilterLink *outlink)
+{
+    ASFSContext *asfs = outlink->src->priv;
+    int ret;
+    int nb_channels =
+        av_get_channel_layout_nb_channels(outlink->channel_layout);
+    int bytes_per_sample_1ch = av_get_bytes_per_sample(outlink->format);
+    int bytes_per_sample = nb_channels * bytes_per_sample_1ch;
+    int bytes_per_outframe = asfs->nb_out_samples * bytes_per_sample;
+    if (av_fifo_size(asfs->fifo) >= bytes_per_outframe)
+        return 1;
+    if (!avfilter_poll_frame(outlink->src->inputs[0]))
+        return 0;
+    ret = avfilter_request_frame(outlink->src->inputs[0]);
+    if (ret < 0) {
+        av_log(outlink->src, AV_LOG_ERROR, "Error occurred when requesting frame from source\n");
+        return ret;
+    }
+    if (av_fifo_size(asfs->fifo) / bytes_per_outframe >= 1)
+        return 1;
+    else
+        return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    ASFSContext *asfs = outlink->src->priv;
+    int nb_channels =
+        av_get_channel_layout_nb_channels(outlink->channel_layout);
+    int bytes_per_sample_1ch = av_get_bytes_per_sample(outlink->format);
+    int bytes_per_sample = nb_channels * bytes_per_sample_1ch;
+    int bytes_per_outframe = asfs->nb_out_samples * bytes_per_sample;
+    AVFilterBufferRef *outsamples;
+
+    if (av_fifo_size(asfs->fifo) < bytes_per_outframe) {
+        av_log(outlink->src, AV_LOG_ERROR, "Requested frame, but having none\n");
+        return AVERROR(EINVAL);
+    }
+
+    outsamples = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, asfs->nb_out_samples);
+    if (!outsamples) {
+        av_log(outlink->src, AV_LOG_ERROR, "avfilter_get_audio_buffer fail\n");
+        return AVERROR(ENOMEM);
+    }
+    av_fifo_generic_read(asfs->fifo, outsamples->data[0], bytes_per_outframe, NULL);
+
+    outsamples->audio->channel_layout = outlink->channel_layout;
+    outsamples->audio->nb_samples = asfs->nb_out_samples;
+    outsamples->audio->sample_rate = outlink->sample_rate;
+    outsamples->pts = asfs->next_out_pts;
+    if (asfs->next_out_pts != AV_NOPTS_VALUE)
+        asfs->next_out_pts += asfs->nb_out_samples;
+
+    avfilter_filter_samples(outlink, outsamples);
+    avfilter_unref_buffer(outsamples);
+    return 0;
+}
+
+AVFilter avfilter_af_asetframesize = {
+    .name           = "asetframesize",
+    .description    = NULL_IF_CONFIG_SMALL(
+            "Set the number of samples for the output audio frames"),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(ASFSContext),
+    .init           = init,
+    .uninit         = uninit,
+
+    .inputs  = (const AVFilterPad[]) {
+        { .name           = "default",
+          .type           = AVMEDIA_TYPE_AUDIO,
+          .filter_samples = filter_samples,
+          .min_perms      = AV_PERM_READ|AV_PERM_WRITE },
+        { .name = NULL }
+    },
+    .outputs = (const AVFilterPad[]) {
+        { .name           = "default",
+          .type           = AVMEDIA_TYPE_AUDIO,
+          .request_frame  = request_frame,
+          .poll_frame     = poll_frame, },
+        { .name = NULL }
+    },
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f0a7f8b..0ed44dc 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -39,6 +39,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (AMERGE,      amerge,      af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
+    REGISTER_FILTER (ASETFRAMESIZE, asetframesize, af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);
     REGISTER_FILTER (ASPLIT,      asplit,      af);
     REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);
-- 
1.7.7



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