示例#1
0
文件: audio.c 项目: SmartJog/ffmpeg
int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
    int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
    AVFilterPad *src = link->srcpad;
    AVFilterPad *dst = link->dstpad;
    int64_t pts;
    AVFilterBufferRef *buf_out;
    int ret;

    FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);

    if (link->closed) {
        avfilter_unref_buffer(samplesref);
        return AVERROR_EOF;
    }

    if (!(filter_samples = dst->filter_samples))
        filter_samples = default_filter_samples;

    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_samples(link, buf_out);
    ff_update_link_current_pts(link, pts);
    return ret;
}
示例#2
0
文件: audio.c 项目: AVbin/libav
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;
}
示例#3
0
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;
}
示例#4
0
void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
    AVFilterPad *dst = link->dstpad;
    int64_t pts;

    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) {
        int  i, size, planar = av_sample_fmt_is_planar(samplesref->format);
        int planes = !planar ? 1:
                     av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);

        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);

        link->cur_buf = ff_default_get_audio_buffer(link, dst->min_perms,
                                                    samplesref->audio->nb_samples);
        link->cur_buf->pts                = samplesref->pts;
        link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;

        /* Copy actual data into new samples buffer */
        /* src can be larger than dst if it was allocated larger than necessary.
           dst can be slightly larger due to extra alignment padding. */
        size = FFMIN(samplesref->linesize[0], link->cur_buf->linesize[0]);
        for (i = 0; samplesref->data[i] && i < 8; i++)
            memcpy(link->cur_buf->data[i], samplesref->data[i], size);
        for (i = 0; i < planes; i++)
            memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], size);

        avfilter_unref_buffer(samplesref);
    } else
        link->cur_buf = samplesref;

    pts = link->cur_buf->pts;
    filter_samples(link, link->cur_buf);
    ff_update_link_current_pts(link, pts);
}
示例#5
0
/* 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->output_count)
        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;
}