@@ -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"

@@ -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

@@ -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

@@ -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);

new file mode 100644

@@ -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}},

+};

new file mode 100644

@@ -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 */

@@ -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, \