/* * 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 "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;} void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref){    ff_filter_samples(link->dst->outputs[0], samplesref);} /* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref){    AVFilterLink *outlink = NULL;     if (inlink->dst->nb_outputs)        outlink = inlink->dst->outputs[0];     if (outlink) {        outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE,                                                       samplesref->audio->nb_samples);        outlink->out_buf->pts                = samplesref->pts;        outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;        ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));        avfilter_unref_buffer(outlink->out_buf);        outlink->out_buf = NULL;    }    avfilter_unref_buffer(samplesref);    inlink->cur_buf = NULL;} void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref){    void (*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 = ff_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;     filter_samples(link, buf_out);}