示例#1
0
static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    MixContext *s      = ctx->priv;
    int i;
    char buf[64];

    s->planar          = av_sample_fmt_is_planar(outlink->format);
    s->sample_rate     = outlink->sample_rate;
#ifdef IDE_COMPILE
	outlink->time_base.num = 1;
	outlink->time_base.den = outlink->sample_rate;
#else
	outlink->time_base = (AVRational){ 1, outlink->sample_rate };
#endif
	s->next_pts        = AV_NOPTS_VALUE;

    s->frame_list = av_mallocz(sizeof(*s->frame_list));
    if (!s->frame_list)
        return AVERROR(ENOMEM);

    s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));
    if (!s->fifos)
        return AVERROR(ENOMEM);

    s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
    for (i = 0; i < s->nb_inputs; i++) {
        s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
        if (!s->fifos[i])
            return AVERROR(ENOMEM);
    }

    s->input_state = av_malloc(s->nb_inputs);
    if (!s->input_state)
        return AVERROR(ENOMEM);
    memset(s->input_state, INPUT_ON, s->nb_inputs);
    s->active_inputs = s->nb_inputs;

    s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
    if (!s->input_scale)
        return AVERROR(ENOMEM);
    s->scale_norm = s->active_inputs;
    calculate_scales(s, 0);

    av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);

    av_log(ctx, AV_LOG_VERBOSE,
           "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);

    return 0;
}
示例#2
0
/**
 * Read samples from the input FIFOs, mix, and write to the output link.
 */
static int output_frame(AVFilterLink *outlink, int nb_samples)
{
    AVFilterContext *ctx = outlink->src;
    MixContext      *s = ctx->priv;
    AVFrame *out_buf, *in_buf;
    int i;

    calculate_scales(s, nb_samples);

    out_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!out_buf)
        return AVERROR(ENOMEM);

    in_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!in_buf) {
        av_frame_free(&out_buf);
        return AVERROR(ENOMEM);
    }

    for (i = 0; i < s->nb_inputs; i++) {
        if (s->input_state[i] == INPUT_ON) {
            int planes, plane_size, p;

            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
                               nb_samples);

            planes     = s->planar ? s->nb_channels : 1;
            plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
            plane_size = FFALIGN(plane_size, 16);

            for (p = 0; p < planes; p++) {
                s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
                                           (float *) in_buf->extended_data[p],
                                           s->input_scale[i], plane_size);
            }
        }
    }
    av_frame_free(&in_buf);

    out_buf->pts = s->next_pts;
    if (s->next_pts != AV_NOPTS_VALUE)
        s->next_pts += nb_samples;

    return ff_filter_frame(outlink, out_buf);
}
示例#3
0
/**
 * Read samples from the input FIFOs, mix, and write to the output link.
 */
static int output_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    MixContext      *s = ctx->priv;
    AVFrame *out_buf, *in_buf;
    int nb_samples, ns, i;

    if (s->input_state[0] & INPUT_ON) {
        /* first input live: use the corresponding frame size */
        nb_samples = frame_list_next_frame_size(s->frame_list);
        for (i = 1; i < s->nb_inputs; i++) {
            if (s->input_state[i] & INPUT_ON) {
                ns = av_audio_fifo_size(s->fifos[i]);
                if (ns < nb_samples) {
                    if (!(s->input_state[i] & INPUT_EOF))
                        /* unclosed input with not enough samples */
                        return 0;
                    /* closed input to drain */
                    nb_samples = ns;
                }
            }
        }
    } else {
        /* first input closed: use the available samples */
        nb_samples = INT_MAX;
        for (i = 1; i < s->nb_inputs; i++) {
            if (s->input_state[i] & INPUT_ON) {
                ns = av_audio_fifo_size(s->fifos[i]);
                nb_samples = FFMIN(nb_samples, ns);
            }
        }
        if (nb_samples == INT_MAX) {
            ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
            return 0;
        }
    }

    s->next_pts = frame_list_next_pts(s->frame_list);
    frame_list_remove_samples(s->frame_list, nb_samples);

    calculate_scales(s, nb_samples);

    if (nb_samples == 0)
        return 0;

    out_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!out_buf)
        return AVERROR(ENOMEM);

    in_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!in_buf) {
        av_frame_free(&out_buf);
        return AVERROR(ENOMEM);
    }

    for (i = 0; i < s->nb_inputs; i++) {
        if (s->input_state[i] & INPUT_ON) {
            int planes, plane_size, p;

            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
                               nb_samples);

            planes     = s->planar ? s->nb_channels : 1;
            plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
            plane_size = FFALIGN(plane_size, 16);

            if (out_buf->format == AV_SAMPLE_FMT_FLT ||
                out_buf->format == AV_SAMPLE_FMT_FLTP) {
                for (p = 0; p < planes; p++) {
                    s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
                                                (float *) in_buf->extended_data[p],
                                                s->input_scale[i], plane_size);
                }
            } else {
                for (p = 0; p < planes; p++) {
                    s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
                                                (double *) in_buf->extended_data[p],
                                                s->input_scale[i], plane_size);
                }
            }
        }
    }
    av_frame_free(&in_buf);

    out_buf->pts = s->next_pts;
    if (s->next_pts != AV_NOPTS_VALUE)
        s->next_pts += nb_samples;

    return ff_filter_frame(outlink, out_buf);
}