ffmpeg/libavfilter/audio.c

177 lines
6.2 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 "libavcodec/avcodec.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
#if FF_API_AVFILTERBUFFER
FF_DISABLE_DEPRECATION_WARNINGS
int avfilter_ref_get_channels(AVFilterBufferRef *ref)
{
return ref->audio ? ref->audio->channels : 0;
}
FF_ENABLE_DEPRECATION_WARNINGS
#endif
AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
return ff_get_audio_buffer(link->dst->outputs[0], nb_samples);
}
AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int channels = link->channels;
int ret;
av_assert0(channels == av_get_channel_layout_nb_channels(link->channel_layout) || !av_get_channel_layout_nb_channels(link->channel_layout));
if (!frame)
return NULL;
frame->nb_samples = nb_samples;
frame->format = link->format;
av_frame_set_channels(frame, link->channels);
frame->channel_layout = link->channel_layout;
frame->sample_rate = link->sample_rate;
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
av_frame_free(&frame);
return NULL;
}
av_samples_set_silence(frame->extended_data, 0, nb_samples, channels,
link->format);
return frame;
}
AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
{
AVFrame *ret = NULL;
if (link->dstpad->get_audio_buffer)
ret = link->dstpad->get_audio_buffer(link, nb_samples);
if (!ret)
ret = ff_default_get_audio_buffer(link, nb_samples);
return ret;
}
#if FF_API_AVFILTERBUFFER
FF_DISABLE_DEPRECATION_WARNINGS
AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_channels(uint8_t **data,
int linesize,int perms,
int nb_samples,
enum AVSampleFormat sample_fmt,
int channels,
uint64_t channel_layout)
{
int planes;
AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
if (!samples || !samplesref)
goto fail;
av_assert0(channels);
av_assert0(channel_layout == 0 ||
channels == av_get_channel_layout_nb_channels(channel_layout));
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_array(sizeof(*samples->extended_data),
planes);
samplesref->extended_data = av_mallocz_array(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;
}
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);
return avfilter_get_audio_buffer_ref_from_arrays_channels(data, linesize, perms,
nb_samples, sample_fmt,
channels, channel_layout);
}
FF_ENABLE_DEPRECATION_WARNINGS
#endif