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/*
* Copyright (c) Stefano Sabatini | stefasab at gmail.com
* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
*
* 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
*/
#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
int nb_samples)
{
return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
}
AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
int nb_samples)
{
AVFilterBufferRef *samplesref = NULL;
uint8_t **data;
int planar = av_sample_fmt_is_planar(link->format);
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
int planes = planar ? nb_channels : 1;
int linesize;
if (!(data = av_mallocz(sizeof(*data) * planes)))
goto fail;
if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
goto fail;
samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
nb_samples, link->format,
link->channel_layout);
if (!samplesref)
goto fail;
av_freep(&data);
fail:
if (data)
av_freep(&data[0]);
av_freep(&data);
return samplesref;
}
AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
int nb_samples)
{
AVFilterBufferRef *ret = NULL;
if (link->dstpad->get_audio_buffer)
ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
if (!ret)
ret = ff_default_get_audio_buffer(link, perms, nb_samples);
if (ret)
ret->type = AVMEDIA_TYPE_AUDIO;
return ret;
}
AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
int linesize,int perms,
int nb_samples,
enum AVSampleFormat sample_fmt,
uint64_t channel_layout)
{
int planes;
AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
if (!samples || !samplesref)
goto fail;
samplesref->buf = samples;
samplesref->buf->free = ff_avfilter_default_free_buffer;
if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
goto fail;
samplesref->audio->nb_samples = nb_samples;
samplesref->audio->channel_layout = channel_layout;
planes = av_sample_fmt_is_planar(sample_fmt) ?
av_get_channel_layout_nb_channels(channel_layout) : 1;
/* make sure the buffer gets read permission or it's useless for output */
samplesref->perms = perms | AV_PERM_READ;
samples->refcount = 1;
samplesref->type = AVMEDIA_TYPE_AUDIO;
samplesref->format = sample_fmt;
memcpy(samples->data, data,
FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
memcpy(samplesref->data, samples->data, sizeof(samples->data));
samples->linesize[0] = samplesref->linesize[0] = linesize;
if (planes > FF_ARRAY_ELEMS(samples->data)) {
samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
planes);
samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
planes);
if (!samples->extended_data || !samplesref->extended_data)
goto fail;
memcpy(samples-> extended_data, data, sizeof(*data)*planes);
memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
} else {
samples->extended_data = samples->data;
samplesref->extended_data = samplesref->data;
}
samplesref->pts = AV_NOPTS_VALUE;
return samplesref;
fail:
if (samples && samples->extended_data != samples->data)
av_freep(&samples->extended_data);
if (samplesref) {
av_freep(&samplesref->audio);
if (samplesref->extended_data != samplesref->data)
av_freep(&samplesref->extended_data);
}
av_freep(&samplesref);
av_freep(&samples);
return NULL;
}
static void default_filter_samples(AVFilterLink *link,
AVFilterBufferRef *samplesref)
{
ff_filter_samples(link->dst->outputs[0], samplesref);
}
void ff_filter_samples_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref)
{
void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
if (!(filter_samples = dst->filter_samples))
filter_samples = default_filter_samples;
/* prepare to copy the samples if the buffer has insufficient permissions */
if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
dst->rej_perms & samplesref->perms) {
av_log(link->dst, AV_LOG_DEBUG,
"Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
samplesref->audio->nb_samples);
buf_out->pts = samplesref->pts;
buf_out->audio->sample_rate = samplesref->audio->sample_rate;
/* Copy actual data into new samples buffer */
av_samples_copy(buf_out->extended_data, samplesref->extended_data,
0, 0, samplesref->audio->nb_samples,
av_get_channel_layout_nb_channels(link->channel_layout),
link->format);
avfilter_unref_buffer(samplesref);
} else
buf_out = samplesref;
link->cur_buf = buf_out;
pts = buf_out->pts;
filter_samples(link, buf_out);
ff_update_link_current_pts(link, pts);
}
void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
AVFilterBufferRef *pbuf = link->partial_buf;
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (!link->min_samples ||
(!pbuf &&
insamples >= link->min_samples && insamples <= link->max_samples)) {
ff_filter_samples_framed(link, samplesref);
return;
}
/* Handle framing (min_samples, max_samples) */
while (insamples) {
if (!pbuf) {
AVRational samples_tb = { 1, link->sample_rate };
int perms = link->dstpad->min_perms | AV_PERM_WRITE;
pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
if (!pbuf) {
av_log(link->dst, AV_LOG_WARNING,
"Samples dropped due to memory allocation failure.\n");
return;
}
avfilter_copy_buffer_ref_props(pbuf, samplesref);
pbuf->pts = samplesref->pts +
av_rescale_q(inpos, samples_tb, link->time_base);
pbuf->audio->nb_samples = 0;
}
nb_samples = FFMIN(insamples,
link->partial_buf_size - pbuf->audio->nb_samples);
av_samples_copy(pbuf->extended_data, samplesref->extended_data,
pbuf->audio->nb_samples, inpos,
nb_samples, nb_channels, link->format);
inpos += nb_samples;
insamples -= nb_samples;
pbuf->audio->nb_samples += nb_samples;
if (pbuf->audio->nb_samples >= link->min_samples) {
ff_filter_samples_framed(link, pbuf);
pbuf = NULL;
}
}
avfilter_unref_buffer(samplesref);
link->partial_buf = pbuf;
}
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