[FFmpeg-devel] [PATCH 3/5] lavfi: add asrc_abuffer - audio source buffer filter
Mina Nagy Zaki
mnzaki at gmail.com
Mon Aug 8 10:11:47 CEST 2011
Originally based on code by Stefano Sabatini and S. N. Hemanth
---
doc/filters.texi | 44 ++++++
libavfilter/Makefile | 2 +
libavfilter/allfilters.c | 1 +
libavfilter/asrc_abuffer.c | 368 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/asrc_abuffer.h | 81 ++++++++++
5 files changed, 496 insertions(+), 0 deletions(-)
create mode 100644 libavfilter/asrc_abuffer.c
create mode 100644 libavfilter/asrc_abuffer.h
diff --git a/doc/filters.texi b/doc/filters.texi
index 8dc1c15..53017b2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -131,6 +131,50 @@ Pass the audio source unchanged to the output.
Below is a description of the currently available audio sources.
+ at section abuffer
+
+Buffer audio frames, and make them available to the filter chain.
+
+This source is mainly intended for a programmatic use, in particular
+through the interface defined in @file{libavfilter/asrc_abuffer.h}.
+
+It accepts the following mandatory parameters:
+ at var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing}
+
+ at table @option
+
+ at item sample_rate
+The sample rate of the incoming audio buffers.
+
+ at item sample_fmt
+The sample format of the incoming audio buffers.
+Either a sample format name or its corresponging integer representation from
+the enum AVSampleFormat in @file{libavutil/samplefmt.h}
+
+ at item channel_layout
+The channel layout of the incoming audio buffers.
+Either a channel layout name from channel_layout_map in
+ at file{libavutil/audioconvert.c} or its corresponding integer representation
+from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h}
+
+ at item packing
+Either "packed" or "planar", or their integer representation: 0 or 1
+respectively.
+
+ at end table
+
+For example:
+ at example
+abuffer=44100:s16:stereo:planar
+ at end example
+
+will instruct the source to accept planar 16bit signed stereo at 44100Hz.
+Since the sample format with name "s16" corresponds to the number
+1 and the "stereo" channel layout corresponds to the value 3
+ at example
+abuffer=44100:1:3:1
+ at end example
+
@section anullsrc
Null audio source, never return audio frames. It is mainly useful as a
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 865ba1e..686fd30 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -65,6 +65,8 @@ OBJS-$(CONFIG_UNSHARP_FILTER) += vf_unsharp.o
OBJS-$(CONFIG_VFLIP_FILTER) += vf_vflip.o
OBJS-$(CONFIG_YADIF_FILTER) += vf_yadif.o
+OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
+
OBJS-$(CONFIG_BUFFER_FILTER) += vsrc_buffer.o
OBJS-$(CONFIG_COLOR_FILTER) += vsrc_color.o
OBJS-$(CONFIG_FREI0R_SRC_FILTER) += vf_frei0r.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 56baa50..49be7b7 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -37,6 +37,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (ANULL, anull, af);
+ REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
REGISTER_FILTER (ANULLSINK, anullsink, asink);
diff --git a/libavfilter/asrc_abuffer.c b/libavfilter/asrc_abuffer.c
new file mode 100644
index 0000000..c19914c
--- /dev/null
+++ b/libavfilter/asrc_abuffer.c
@@ -0,0 +1,368 @@
+/*
+ * Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * 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
+ * Memory buffer source filter for audio
+ */
+
+#include "libavutil/audioconvert.h"
+#include "asrc_abuffer.h"
+#include "internal.h"
+
+typedef struct {
+ // Audio format of incoming buffers
+ int sample_rate;
+ unsigned int sample_fmt;
+ int64_t channel_layout;
+ int planar;
+ // FIFO buffer of audio buffer ref pointers
+ AVFifoBuffer *fifo;
+ // Normalization filters
+ AVFilterContext *aconvert;
+ AVFilterContext *aresample;
+} ABufferSourceContext;
+
+#define FIFO_SIZE 8
+
+static void buf_free(AVFilterBuffer *ptr)
+{
+ av_free(ptr);
+ return;
+}
+
+static void set_link_source(AVFilterContext *src, AVFilterLink *link)
+{
+ link->src = src;
+ link->srcpad = &(src->output_pads[0]);
+ src->outputs[0] = link;
+}
+
+static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx)
+{
+ int ret;
+ AVFilterLink * const inlink = filt_ctx->inputs[0];
+ AVFilterLink * const outlink = filt_ctx->outputs[0];
+
+ inlink->format = abuffer->sample_fmt;
+ inlink->channel_layout = abuffer->channel_layout;
+ inlink->planar = abuffer->planar;
+ inlink->sample_rate = abuffer->sample_rate;
+
+ filt_ctx->filter->uninit(filt_ctx);
+ memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size);
+ if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0)
+ return ret;
+ if ((ret = inlink->srcpad->config_props(inlink)) < 0)
+ return ret;
+ return outlink->srcpad->config_props(outlink);
+}
+
+static int insert_filter(ABufferSourceContext *abuffer,
+ AVFilterLink *link, AVFilterContext **filt_ctx,
+ const char *filt_name)
+{
+ int ret;
+
+ if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0)
+ return ret;
+
+ link->src->outputs[0] = NULL;
+ if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) {
+ link->src->outputs[0] = link;
+ return ret;
+ }
+
+ set_link_source(*filt_ctx, link);
+
+ if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) {
+ avfilter_free(*filt_ctx);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void remove_filter(AVFilterContext **filt_ctx)
+{
+ AVFilterLink *outlink = (*filt_ctx)->outputs[0];
+ AVFilterContext *src = (*filt_ctx)->inputs[0]->src;
+
+ (*filt_ctx)->outputs[0] = NULL;
+ avfilter_free(*filt_ctx);
+ *filt_ctx = NULL;
+
+ set_link_source(src, outlink);
+}
+
+static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref)
+{
+ char old_layout_str[16], new_layout_str[16];
+ av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str),
+ -1, link->channel_layout);
+ av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str),
+ -1, ref->audio->channel_layout);
+ av_log(ctx, AV_LOG_INFO,
+ "Audio input format changed: "
+ "%s:%s:%u -> %s:%s:%u, normalizing\n",
+ av_get_sample_fmt_name(link->format),
+ old_layout_str, link->sample_rate,
+ av_get_sample_fmt_name(ref->format),
+ new_layout_str, ref->audio->sample_rate);
+}
+
+int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx,
+ AVFilterBufferRef *samplesref,
+ int av_unused flags)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterLink *link;
+ int ret, logged = 0;
+
+ if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Buffering limit reached. Please consume some available frames "
+ "before adding new ones.\n");
+ return AVERROR(EINVAL);
+ }
+
+ // Normalize input
+
+ link = ctx->outputs[0];
+ if (samplesref->audio->sample_rate != link->sample_rate) {
+
+ log_input_change(ctx, link, samplesref);
+ logged = 1;
+
+ abuffer->sample_rate = samplesref->audio->sample_rate;
+
+ if (!abuffer->aresample) {
+ ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aresample->outputs[0];
+ if (samplesref->audio->sample_rate == link->sample_rate)
+ remove_filter(&abuffer->aresample);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0)
+ return ret;
+ }
+ }
+
+ link = ctx->outputs[0];
+ if (samplesref->format != link->format ||
+ samplesref->audio->channel_layout != link->channel_layout ||
+ samplesref->audio->planar != link->planar) {
+
+ if (!logged) log_input_change(ctx, link, samplesref);
+
+ abuffer->sample_fmt = samplesref->format;
+ abuffer->channel_layout = samplesref->audio->channel_layout;
+ abuffer->planar = samplesref->audio->planar;
+
+ if (!abuffer->aconvert) {
+ ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert");
+ if (ret < 0) return ret;
+ } else {
+ link = abuffer->aconvert->outputs[0];
+ if (samplesref->format == link->format &&
+ samplesref->audio->channel_layout == link->channel_layout &&
+ samplesref->audio->planar == link->planar
+ )
+ remove_filter(&abuffer->aconvert);
+ else
+ if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0)
+ return ret;
+ }
+ }
+
+ if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref,
+ sizeof(samplesref), NULL)) {
+ av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n");
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+int av_asrc_buffer_add_samples(AVFilterContext *ctx,
+ uint8_t *data[8], int linesize[8],
+ int nb_samples, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ AVFilterBufferRef *samplesref;
+
+ samplesref = avfilter_get_audio_buffer_ref_from_arrays(
+ data, linesize, AV_PERM_WRITE,
+ nb_samples,
+ sample_fmt, channel_layout, planar);
+ if (!samplesref)
+ return AVERROR(ENOMEM);
+
+ samplesref->buf->free = buf_free;
+ samplesref->pts = pts;
+ samplesref->audio->sample_rate = sample_rate;
+
+ return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0);
+}
+
+int av_asrc_buffer_add_buffer(AVFilterContext *ctx,
+ uint8_t *buf, int buf_size, int sample_rate,
+ int sample_fmt, int64_t channel_layout, int planar,
+ int64_t pts, int av_unused flags)
+{
+ uint8_t *data[8];
+ int linesize[8];
+ int nb_channels = av_get_channel_layout_nb_channels(channel_layout),
+ nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt);
+
+ av_samples_fill_arrays(data, linesize,
+ buf, nb_channels, nb_samples,
+ sample_fmt, planar, 16);
+
+ return av_asrc_buffer_add_samples(ctx,
+ data, linesize, nb_samples,
+ sample_rate,
+ sample_fmt, channel_layout, planar,
+ pts, flags);
+}
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ char *arg, chlayout_str[16];
+
+ arg = strsep(&args, ":");
+ if (!arg) goto arg_fail;
+ abuffer->sample_rate = ff_parse_sample_rate(arg, ctx);
+ if (abuffer->sample_rate == -1) return AVERROR(EINVAL);
+
+ arg = strsep(&args, ":");
+ if (!arg) goto arg_fail;
+ abuffer->sample_fmt = ff_parse_sample_format(arg, ctx);
+ if (abuffer->sample_fmt == -1) return AVERROR(EINVAL);
+
+ arg = strsep(&args, ":");
+ if (!arg) goto arg_fail;
+ abuffer->channel_layout = ff_parse_channel_layout(arg, ctx);
+ if (abuffer->channel_layout == -1) return AVERROR(EINVAL);
+
+ arg = strsep(&args, ":");
+ if (!arg) goto arg_fail;
+ abuffer->planar = ff_parse_packing_format(arg, ctx);
+ if (abuffer->planar == -1) return AVERROR(EINVAL);
+
+ abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*));
+ if (!abuffer->fifo) {
+ av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n");
+ return AVERROR(ENOMEM);
+ }
+
+ av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str),
+ -1, abuffer->channel_layout);
+ av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n",
+ av_get_sample_fmt_name(abuffer->sample_fmt), chlayout_str,
+ abuffer->sample_rate);
+
+ return 0;
+
+arg_fail:
+ av_log(ctx, AV_LOG_ERROR, "Please provide the required arguments "
+ "sample_rate:sample_fmt:channel_layout:packing\n");
+ return AVERROR(EINVAL);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ av_fifo_free(abuffer->fifo);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ ABufferSourceContext *abuffer = ctx->priv;
+ AVFilterFormats *formats;
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->sample_fmt);
+ avfilter_set_common_sample_formats(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats, abuffer->channel_layout);
+ avfilter_set_common_channel_layouts(ctx, formats);
+
+ formats = NULL;
+ avfilter_add_format(&formats,
+ abuffer->planar ? AVFILTER_PLANAR : AVFILTER_PACKED);
+ avfilter_set_common_packing_formats(ctx, formats);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ outlink->sample_rate = abuffer->sample_rate;
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ ABufferSourceContext *abuffer = outlink->src->priv;
+ AVFilterBufferRef *samplesref;
+
+ if (!av_fifo_size(abuffer->fifo)) {
+ av_log(outlink->src, AV_LOG_ERROR,
+ "request_frame() called with no available frames!\n");
+ return AVERROR(EINVAL);
+ }
+
+ av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL);
+ avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
+ avfilter_unref_buffer(samplesref);
+
+ return 0;
+}
+
+static int poll_frame(AVFilterLink *link)
+{
+ ABufferSourceContext *abuffer = link->src->priv;
+ return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*);
+}
+
+AVFilter avfilter_asrc_abuffer = {
+ .name = "abuffer",
+ .description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
+ .priv_size = sizeof(ABufferSourceContext),
+ .query_formats = query_formats,
+
+ .init = init,
+ .uninit = uninit,
+
+ .inputs = (AVFilterPad[]) {{ .name = NULL }},
+ .outputs = (AVFilterPad[]) {{ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .poll_frame = poll_frame,
+ .config_props = config_output, },
+ { .name = NULL}},
+};
diff --git a/libavfilter/asrc_abuffer.h b/libavfilter/asrc_abuffer.h
new file mode 100644
index 0000000..1f1d9e0
--- /dev/null
+++ b/libavfilter/asrc_abuffer.h
@@ -0,0 +1,81 @@
+/*
+ * 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
+ */
+
+#ifndef AVFILTER_ASRC_ABUFFER_H
+#define AVFILTER_ASRC_ABUFFER_H
+
+#include "libavutil/fifo.h"
+#include "avfilter.h"
+
+/**
+ * @file
+ * memory buffer source filter for audio
+ */
+
+/**
+ * Queue an audio buffer to the audio buffer source.
+ *
+ * @param abuffersrc audio source buffer context
+ * @param data pointers to the samples planes
+ * @param linesize linesizes of each audio buffer plane
+ * @param nb_samples number of samples per channel
+ * @param sample_fmt sample format of the audio data
+ * @param ch_layout channel layout of the audio data
+ * @param planar flag to indicate if audio data is planar or packed
+ * @param pts presentation timestamp of the audio buffer
+ * @param flags unused
+ */
+int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc,
+ uint8_t *data[8], int linesize[8],
+ int nb_samples, int sample_rate,
+ int sample_fmt, int64_t ch_layout, int planar,
+ int64_t pts, int av_unused flags);
+
+/**
+ * Queue an audio buffer to the audio buffer source.
+ *
+ * This is similar to av_asrc_buffer_add_samples(), but the samples
+ * are stored in a buffer with known size.
+ *
+ * @param abuffersrc audio source buffer context
+ * @param buf pointer to the samples data, packed is assumed
+ * @param size the size in bytes of the buffer, it must contain an
+ * integer number of samples
+ * @param sample_fmt sample format of the audio data
+ * @param ch_layout channel layout of the audio data
+ * @param pts presentation timestamp of the audio buffer
+ * @param flags unused
+ */
+int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc,
+ uint8_t *buf, int buf_size,
+ int sample_rate,
+ int sample_fmt, int64_t ch_layout, int planar,
+ int64_t pts, int av_unused flags);
+
+/**
+ * Queue an audio buffer to the audio buffer source.
+ *
+ * @param abuffersrc audio source buffer context
+ * @param samplesref buffer ref to queue
+ * @param flags unused
+ */
+int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *abuffersrc,
+ AVFilterBufferRef *samplesref,
+ int av_unused flags);
+
+#endif /* AVFILTER_ASRC_ABUFFER_H */
--
1.7.4.4
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