lavfi: add asrc_abuffer - audio buffer source
authorMina Nagy Zaki <mnzaki@gmail.com>
Mon, 1 Aug 2011 08:33:26 +0000 (11:33 +0300)
committerStefano Sabatini <stefano.sabatini-lala@poste.it>
Sun, 21 Aug 2011 09:37:57 +0000 (11:37 +0200)
Originally based on code by Stefano Sabatini and S. N. Hemanth.

Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
configure
doc/filters.texi
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/asrc_abuffer.c [new file with mode: 0644]
libavfilter/asrc_abuffer.h [new file with mode: 0644]
libavfilter/avfilter.h

index 26f6d21..9cee2a0 100755 (executable)
--- a/configure
+++ b/configure
@@ -1501,6 +1501,7 @@ tcp_protocol_deps="network"
 udp_protocol_deps="network"
 
 # filters
+abuffer="strtok_r"
 aformat_filter_deps="strtok_r"
 blackframe_filter_deps="gpl"
 boxblur_filter_deps="gpl"
index dd99c73..69ba4b1 100644 (file)
@@ -194,6 +194,51 @@ Adler-32 checksum for each input frame plane, expressed in the form
 
 Below is a description of the currently available audio sources.
 
+@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:
+@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing}
+
+@table @option
+
+@item sample_rate
+The sample rate of the incoming audio buffers.
+
+@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}
+
+@item channel_layout
+The channel layout of the incoming audio buffers.
+Either a channel layout name from channel_layout_map in
+@file{libavutil/audioconvert.c} or its corresponding integer representation
+from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h}
+
+@item packing
+Either "packed" or "planar", or their integer representation: 0 or 1
+respectively.
+
+@end table
+
+For example:
+@example
+abuffer=44100:s16:stereo:planar
+@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, this is
+equivalent to:
+@example
+abuffer=44100:1:3:1
+@end example
+
 @section anullsrc
 
 Null audio source, never return audio frames. It is mainly useful as a
index a3faf27..5ed7e99 100644 (file)
@@ -24,6 +24,7 @@ OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
 
+OBJS-$(CONFIG_ABUFFER_FILTER)                += asrc_abuffer.o
 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o
 
 OBJS-$(CONFIG_ABUFFERSINK_FILTER)            += asink_abuffer.o
index 5cf330c..3675561 100644 (file)
@@ -39,6 +39,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);
 
+    REGISTER_FILTER (ABUFFER,     abuffer,     asrc);
     REGISTER_FILTER (ANULLSRC,    anullsrc,    asrc);
 
     REGISTER_FILTER (ABUFFERSINK, abuffersink, asink);
diff --git a/libavfilter/asrc_abuffer.c b/libavfilter/asrc_abuffer.c
new file mode 100644 (file)
index 0000000..badc2d8
--- /dev/null
@@ -0,0 +1,366 @@
+/*
+ * 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 for audio
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/fifo.h"
+#include "asrc_abuffer.h"
+#include "internal.h"
+
+typedef struct {
+    // Audio format of incoming buffers
+    int sample_rate;
+    unsigned int sample_format;
+    int64_t channel_layout;
+    int packing_format;
+
+    // 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_format;
+    inlink->channel_layout = abuffer->channel_layout;
+    inlink->planar         = abuffer->packing_format;
+    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:%"PRId64" -> %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_format  = samplesref->format;
+        abuffer->channel_layout = samplesref->audio->channel_layout;
+        abuffer->packing_format = 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 = NULL, *ptr, chlayout_str[16];
+    int ret;
+
+    arg = strtok_r(args, ":", &ptr);
+
+#define ADD_FORMAT(fmt_name)                                            \
+    if (!arg)                                                           \
+        goto arg_fail;                                                  \
+    if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0)  \
+        return ret;                                                     \
+    if (*args)                                                          \
+        arg = strtok_r(NULL, ":", &ptr)
+
+    ADD_FORMAT(sample_rate);
+    ADD_FORMAT(sample_format);
+    ADD_FORMAT(channel_layout);
+    ADD_FORMAT(packing_format);
+
+    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_format), chlayout_str,
+           abuffer->sample_rate);
+
+    return 0;
+
+arg_fail:
+    av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form "
+                              "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_format);
+    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->packing_format);
+    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 *outlink)
+{
+    ABufferSourceContext *abuffer = outlink->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 (file)
index 0000000..4352c74
--- /dev/null
@@ -0,0 +1,80 @@
+/*
+ * 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 "avfilter.h"
+
+/**
+ * @file
+ * memory buffer source 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 */
index af2d3cd..68b5d23 100644 (file)
@@ -29,7 +29,7 @@
 #include "libavutil/rational.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 33
+#define LIBAVFILTER_VERSION_MINOR 34
 #define LIBAVFILTER_VERSION_MICRO  0
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \