float mag_total,
float x, float y,
int n);
+ void (*upmix_5_0)(AVFilterContext *ctx,
+ float c_re, float c_im,
+ float mag_totall, float mag_totalr,
+ float fl_phase, float fr_phase,
+ float bl_phase, float br_phase,
+ float sl_phase, float sr_phase,
+ float xl, float yl,
+ float xr, float yr,
+ int n);
void (*upmix_5_1)(AVFilterContext *ctx,
float c_re, float c_im,
float lfe_re, float lfe_im,
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
+static void upmix_7_1_5_0_side(AVFilterContext *ctx,
+ float c_re, float c_im,
+ float mag_totall, float mag_totalr,
+ float fl_phase, float fr_phase,
+ float bl_phase, float br_phase,
+ float sl_phase, float sr_phase,
+ float xl, float yl,
+ float xr, float yr,
+ int n)
+{
+ float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
+ float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
+ float lfe_mag, c_phase, mag_total = (mag_totall + mag_totalr) * 0.5;
+ AudioSurroundContext *s = ctx->priv;
+
+ dstl = (float *)s->output->extended_data[0];
+ dstr = (float *)s->output->extended_data[1];
+ dstc = (float *)s->output->extended_data[2];
+ dstlfe = (float *)s->output->extended_data[3];
+ dstlb = (float *)s->output->extended_data[4];
+ dstrb = (float *)s->output->extended_data[5];
+ dstls = (float *)s->output->extended_data[6];
+ dstrs = (float *)s->output->extended_data[7];
+
+ c_phase = atan2f(c_im, c_re);
+
+ get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
+
+ fl_mag = sqrtf(.5f * (xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
+ fr_mag = sqrtf(.5f * (xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
+ lb_mag = sqrtf(.5f * (-xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
+ rb_mag = sqrtf(.5f * (-xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
+ ls_mag = sqrtf(1.f - fabsf(xl)) * ((yl + 1.f) * .5f) * mag_totall;
+ rs_mag = sqrtf(1.f - fabsf(xr)) * ((yr + 1.f) * .5f) * mag_totalr;
+
+ dstl[2 * n ] = fl_mag * cosf(fl_phase);
+ dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
+
+ dstr[2 * n ] = fr_mag * cosf(fr_phase);
+ dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
+
+ dstc[2 * n ] = c_re;
+ dstc[2 * n + 1] = c_im;
+
+ dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
+ dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+
+ dstlb[2 * n ] = lb_mag * cosf(bl_phase);
+ dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
+
+ dstrb[2 * n ] = rb_mag * cosf(br_phase);
+ dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
+
+ dstls[2 * n ] = ls_mag * cosf(sl_phase);
+ dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
+
+ dstrs[2 * n ] = rs_mag * cosf(sr_phase);
+ dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
+}
+
static void upmix_7_1_5_1(AVFilterContext *ctx,
float c_re, float c_im,
float lfe_re, float lfe_im,
}
}
+static void filter_5_0_side(AVFilterContext *ctx)
+{
+ AudioSurroundContext *s = ctx->priv;
+ float *srcl, *srcr, *srcc, *srcsl, *srcsr;
+ int n;
+
+ srcl = (float *)s->input->extended_data[0];
+ srcr = (float *)s->input->extended_data[1];
+ srcc = (float *)s->input->extended_data[2];
+ srcsl = (float *)s->input->extended_data[3];
+ srcsr = (float *)s->input->extended_data[4];
+
+ for (n = 0; n < s->buf_size; n++) {
+ float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
+ float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
+ float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
+ float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
+ float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
+ float fl_mag = hypotf(fl_re, fl_im);
+ float fr_mag = hypotf(fr_re, fr_im);
+ float fl_phase = atan2f(fl_im, fl_re);
+ float fr_phase = atan2f(fr_im, fr_re);
+ float sl_mag = hypotf(sl_re, sl_im);
+ float sr_mag = hypotf(sr_re, sr_im);
+ float sl_phase = atan2f(sl_im, sl_re);
+ float sr_phase = atan2f(sr_im, sr_re);
+ float phase_difl = fabsf(fl_phase - sl_phase);
+ float phase_difr = fabsf(fr_phase - sr_phase);
+ float mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
+ float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
+ float mag_totall = hypotf(fl_mag, sl_mag);
+ float mag_totalr = hypotf(fr_mag, sr_mag);
+ float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
+ float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
+ float xl, yl;
+ float xr, yr;
+
+ if (phase_difl > M_PI)
+ phase_difl = 2 * M_PI - phase_difl;
+
+ if (phase_difr > M_PI)
+ phase_difr = 2 * M_PI - phase_difr;
+
+ stereo_position(mag_difl, phase_difl, &xl, &yl);
+ stereo_position(mag_difr, phase_difr, &xr, &yr);
+
+ s->upmix_5_0(ctx, c_re, c_im,
+ mag_totall, mag_totalr,
+ fl_phase, fr_phase,
+ bl_phase, br_phase,
+ sl_phase, sr_phase,
+ xl, yl, xr, yr, n);
+ }
+}
+
+static void filter_5_1_side(AVFilterContext *ctx)
+{
+ AudioSurroundContext *s = ctx->priv;
+ float *srcl, *srcr, *srcc, *srclfe, *srcsl, *srcsr;
+ int n;
+
+ srcl = (float *)s->input->extended_data[0];
+ srcr = (float *)s->input->extended_data[1];
+ srcc = (float *)s->input->extended_data[2];
+ srclfe = (float *)s->input->extended_data[3];
+ srcsl = (float *)s->input->extended_data[4];
+ srcsr = (float *)s->input->extended_data[5];
+
+ for (n = 0; n < s->buf_size; n++) {
+ float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
+ float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
+ float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
+ float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
+ float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
+ float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
+ float fl_mag = hypotf(fl_re, fl_im);
+ float fr_mag = hypotf(fr_re, fr_im);
+ float fl_phase = atan2f(fl_im, fl_re);
+ float fr_phase = atan2f(fr_im, fr_re);
+ float sl_mag = hypotf(sl_re, sl_im);
+ float sr_mag = hypotf(sr_re, sr_im);
+ float sl_phase = atan2f(sl_im, sl_re);
+ float sr_phase = atan2f(sr_im, sr_re);
+ float phase_difl = fabsf(fl_phase - sl_phase);
+ float phase_difr = fabsf(fr_phase - sr_phase);
+ float mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
+ float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
+ float mag_totall = hypotf(fl_mag, sl_mag);
+ float mag_totalr = hypotf(fr_mag, sr_mag);
+ float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
+ float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
+ float xl, yl;
+ float xr, yr;
+
+ if (phase_difl > M_PI)
+ phase_difl = 2 * M_PI - phase_difl;
+
+ if (phase_difr > M_PI)
+ phase_difr = 2 * M_PI - phase_difr;
+
+ stereo_position(mag_difl, phase_difl, &xl, &yl);
+ stereo_position(mag_difr, phase_difr, &xr, &yr);
+
+ s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
+ mag_totall, mag_totalr,
+ fl_phase, fr_phase,
+ bl_phase, br_phase,
+ sl_phase, sr_phase,
+ xl, yl, xr, yr, n);
+ }
+}
+
static void filter_5_1_back(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
goto fail;
}
break;
+ case AV_CH_LAYOUT_5POINT0:
+ s->filter = filter_5_0_side;
+ switch (s->out_channel_layout) {
+ case AV_CH_LAYOUT_7POINT1:
+ s->upmix_5_0 = upmix_7_1_5_0_side;
+ break;
+ default:
+ goto fail;
+ }
+ break;
+ case AV_CH_LAYOUT_5POINT1:
+ s->filter = filter_5_1_side;
+ switch (s->out_channel_layout) {
+ case AV_CH_LAYOUT_7POINT1:
+ s->upmix_5_1 = upmix_7_1_5_1;
+ break;
+ default:
+ goto fail;
+ }
+ break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->filter = filter_5_1_back;
switch (s->out_channel_layout) {
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
+ int ret;
- av_audio_fifo_write(s->fifo, (void **)in->extended_data,
- in->nb_samples);
-
- if (s->pts == AV_NOPTS_VALUE)
+ ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+ in->nb_samples);
+ if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
av_frame_free(&in);
+ if (ret < 0)
+ return ret;
while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
AVFrame *out;
- int ret;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
if (ret < 0)
return 0;
}
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioSurroundContext *s = ctx->priv;
+ int ret = 0;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0 && av_audio_fifo_size(s->fifo) < s->buf_size) {
+ AVFrame *in;
+
+ in = ff_get_audio_buffer(outlink, s->buf_size - av_audio_fifo_size(s->fifo));
+ if (!in)
+ return AVERROR(ENOMEM);
+ ret = filter_frame(ctx->inputs[0], in);
+ av_audio_fifo_drain(s->fifo, s->buf_size);
+ }
+
+ return ret;
+}
+
static av_cold void uninit(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
static const AVFilterPad outputs[] = {
{
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .config_props = config_output,
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_output,
},
{ NULL }
};
projects / ffmpeg / blobdiff