mirror of https://git.ffmpeg.org/ffmpeg.git
255 lines
9.1 KiB
C
255 lines
9.1 KiB
C
/*
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* Copyright (c) Stefano Sabatini | stefasab at gmail.com
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* Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/audioconvert.h"
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#include "libavutil/common.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples)
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{
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return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
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}
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AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples)
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{
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AVFilterBufferRef *samplesref = NULL;
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uint8_t **data;
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int planar = av_sample_fmt_is_planar(link->format);
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int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
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int planes = planar ? nb_channels : 1;
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int linesize;
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if (!(data = av_mallocz(sizeof(*data) * planes)))
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goto fail;
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if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
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goto fail;
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samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
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nb_samples, link->format,
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link->channel_layout);
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if (!samplesref)
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goto fail;
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av_freep(&data);
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fail:
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if (data)
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av_freep(&data[0]);
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av_freep(&data);
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return samplesref;
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}
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AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
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int nb_samples)
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{
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AVFilterBufferRef *ret = NULL;
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if (link->dstpad->get_audio_buffer)
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ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
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if (!ret)
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ret = ff_default_get_audio_buffer(link, perms, nb_samples);
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if (ret)
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ret->type = AVMEDIA_TYPE_AUDIO;
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return ret;
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}
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AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
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int linesize,int perms,
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int nb_samples,
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enum AVSampleFormat sample_fmt,
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uint64_t channel_layout)
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{
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int planes;
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AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
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AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
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if (!samples || !samplesref)
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goto fail;
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samplesref->buf = samples;
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samplesref->buf->free = ff_avfilter_default_free_buffer;
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if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
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goto fail;
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samplesref->audio->nb_samples = nb_samples;
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samplesref->audio->channel_layout = channel_layout;
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planes = av_sample_fmt_is_planar(sample_fmt) ?
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av_get_channel_layout_nb_channels(channel_layout) : 1;
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/* make sure the buffer gets read permission or it's useless for output */
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samplesref->perms = perms | AV_PERM_READ;
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samples->refcount = 1;
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samplesref->type = AVMEDIA_TYPE_AUDIO;
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samplesref->format = sample_fmt;
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memcpy(samples->data, data,
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FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
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memcpy(samplesref->data, samples->data, sizeof(samples->data));
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samples->linesize[0] = samplesref->linesize[0] = linesize;
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if (planes > FF_ARRAY_ELEMS(samples->data)) {
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samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
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planes);
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samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
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planes);
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if (!samples->extended_data || !samplesref->extended_data)
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goto fail;
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memcpy(samples-> extended_data, data, sizeof(*data)*planes);
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memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
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} else {
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samples->extended_data = samples->data;
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samplesref->extended_data = samplesref->data;
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}
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samplesref->pts = AV_NOPTS_VALUE;
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return samplesref;
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fail:
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if (samples && samples->extended_data != samples->data)
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av_freep(&samples->extended_data);
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if (samplesref) {
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av_freep(&samplesref->audio);
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if (samplesref->extended_data != samplesref->data)
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av_freep(&samplesref->extended_data);
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}
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av_freep(&samplesref);
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av_freep(&samples);
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return NULL;
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}
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static int default_filter_samples(AVFilterLink *link,
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AVFilterBufferRef *samplesref)
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{
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return ff_filter_samples(link->dst->outputs[0], samplesref);
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}
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int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
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{
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int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
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AVFilterPad *src = link->srcpad;
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AVFilterPad *dst = link->dstpad;
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int64_t pts;
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AVFilterBufferRef *buf_out;
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int ret;
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FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
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if (!(filter_samples = dst->filter_samples))
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filter_samples = default_filter_samples;
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av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
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samplesref->perms &= ~ src->rej_perms;
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/* prepare to copy the samples if the buffer has insufficient permissions */
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if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
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dst->rej_perms & samplesref->perms) {
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av_log(link->dst, AV_LOG_DEBUG,
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"Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
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samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
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buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
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samplesref->audio->nb_samples);
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if (!buf_out) {
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avfilter_unref_buffer(samplesref);
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return AVERROR(ENOMEM);
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}
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buf_out->pts = samplesref->pts;
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buf_out->audio->sample_rate = samplesref->audio->sample_rate;
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/* Copy actual data into new samples buffer */
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av_samples_copy(buf_out->extended_data, samplesref->extended_data,
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0, 0, samplesref->audio->nb_samples,
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av_get_channel_layout_nb_channels(link->channel_layout),
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link->format);
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avfilter_unref_buffer(samplesref);
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} else
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buf_out = samplesref;
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link->cur_buf = buf_out;
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pts = buf_out->pts;
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ret = filter_samples(link, buf_out);
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ff_update_link_current_pts(link, pts);
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return ret;
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}
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int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
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{
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int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
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AVFilterBufferRef *pbuf = link->partial_buf;
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int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
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int ret = 0;
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if (!link->min_samples ||
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(!pbuf &&
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insamples >= link->min_samples && insamples <= link->max_samples)) {
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return ff_filter_samples_framed(link, samplesref);
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}
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/* Handle framing (min_samples, max_samples) */
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while (insamples) {
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if (!pbuf) {
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AVRational samples_tb = { 1, link->sample_rate };
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int perms = link->dstpad->min_perms | AV_PERM_WRITE;
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pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
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if (!pbuf) {
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av_log(link->dst, AV_LOG_WARNING,
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"Samples dropped due to memory allocation failure.\n");
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return 0;
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}
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avfilter_copy_buffer_ref_props(pbuf, samplesref);
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pbuf->pts = samplesref->pts +
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av_rescale_q(inpos, samples_tb, link->time_base);
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pbuf->audio->nb_samples = 0;
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}
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nb_samples = FFMIN(insamples,
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link->partial_buf_size - pbuf->audio->nb_samples);
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av_samples_copy(pbuf->extended_data, samplesref->extended_data,
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pbuf->audio->nb_samples, inpos,
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nb_samples, nb_channels, link->format);
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inpos += nb_samples;
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insamples -= nb_samples;
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pbuf->audio->nb_samples += nb_samples;
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if (pbuf->audio->nb_samples >= link->min_samples) {
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ret = ff_filter_samples_framed(link, pbuf);
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pbuf = NULL;
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}
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}
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avfilter_unref_buffer(samplesref);
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link->partial_buf = pbuf;
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return ret;
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}
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