/* * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/audioconvert.h" #include "libavutil/common.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; samplesref->audio->planar = av_sample_fmt_is_planar(sample_fmt); planes = samplesref->audio->planar ? 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 int default_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { return ff_filter_samples(link->dst->outputs[0], samplesref); } int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *); AVFilterPad *dst = link->dstpad; AVFilterBufferRef *buf_out; FF_DPRINTF_START(NULL, filter_samples); ff_dlog_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); 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; return filter_samples(link, buf_out); }