ffmpeg/libavfilter/audio.c

276 lines
9.9 KiB
C

/*
* 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/channel_layout.h"
#include "libavutil/common.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
int avfilter_ref_get_channels(AVFilterBufferRef *ref)
{
return ref->audio ? ref->audio->channels : 0;
}
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;
int full_perms = AV_PERM_READ | AV_PERM_WRITE | AV_PERM_PRESERVE |
AV_PERM_REUSE | AV_PERM_REUSE2 | AV_PERM_ALIGN;
av_assert1(!(perms & ~(full_perms | AV_PERM_NEG_LINESIZES)));
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, full_perms,
nb_samples, link->format,
link->channel_layout);
if (!samplesref)
goto fail;
samplesref->audio->sample_rate = link->sample_rate;
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 channels = av_get_channel_layout_nb_channels(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;
samplesref->audio->channels = channels;
planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 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 int default_filter_frame(AVFilterLink *link, AVFilterBufferRef *frame)
{
return ff_filter_frame(link->dst->outputs[0], frame);
}
int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int (*filter_frame)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *src = link->srcpad;
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
int ret;
FF_TPRINTF_START(NULL, filter_frame); ff_tlog_link(NULL, link, 1);
if (link->closed) {
avfilter_unref_buffer(samplesref);
return AVERROR_EOF;
}
if (!(filter_frame = dst->filter_frame))
filter_frame = default_filter_frame;
av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
samplesref->perms &= ~ src->rej_perms;
/* 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);
if (!buf_out) {
avfilter_unref_buffer(samplesref);
return AVERROR(ENOMEM);
}
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;
ret = filter_frame(link, buf_out);
ff_update_link_current_pts(link, pts);
return ret;
}
int 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);
int ret = 0;
av_assert1(samplesref->format == link->format);
av_assert1(samplesref->audio->channels == link->channels);
av_assert1(samplesref->audio->channel_layout == link->channel_layout);
av_assert1(samplesref->audio->sample_rate == link->sample_rate);
if (!link->min_samples ||
(!pbuf &&
insamples >= link->min_samples && insamples <= link->max_samples)) {
return ff_filter_samples_framed(link, samplesref);
}
/* 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 0;
}
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) {
ret = ff_filter_samples_framed(link, pbuf);
pbuf = NULL;
}
}
avfilter_unref_buffer(samplesref);
link->partial_buf = pbuf;
return ret;
}