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avfilter/af_surround: do not rewrite pts any more 

Also stop using fifo and excessive peeking.
This commit is contained in:
Paul B Mahol 2022-02-22 13:02:41 +01:00
parent c337b0f826
commit fee804f7ed
1 changed files with 32 additions and 65 deletions
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97
libavfilter/af_surround.c
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@ -19,7 +19,6 @@
*/
#include "libavutil/avassert.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
@ -102,17 +101,14 @@ typedef struct AudioSurroundContext {
AVFrame *output;
AVFrame *output_out;
AVFrame *overlap_buffer;
AVFrame *window;
int buf_size;
int hop_size;
AVAudioFifo *fifo;
AVTXContext **rdft, **irdft;
av_tx_fn tx_fn, itx_fn;
float *window_func_lut;
int64_t pts;
int eof;
void (*filter)(AVFilterContext *ctx);
void (*upmix_stereo)(AVFilterContext *ctx,
float l_phase,
@ -245,7 +241,11 @@ static int config_input(AVFilterLink *inlink)
if (ch >= 0)
s->input_levels[ch] *= s->lfe_in;
s->input_in = ff_get_audio_buffer(inlink, s->buf_size + 2);
s->window = ff_get_audio_buffer(inlink, s->buf_size * 2);
if (!s->window)
return AVERROR(ENOMEM);
s->input_in = ff_get_audio_buffer(inlink, s->buf_size * 2);
if (!s->input_in)
return AVERROR(ENOMEM);
@ -253,10 +253,6 @@ static int config_input(AVFilterLink *inlink)
if (!s->input)
return AVERROR(ENOMEM);
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->buf_size);
if (!s->fifo)
return AVERROR(ENOMEM);
s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
@ -1513,7 +1509,6 @@ fail:
}
s->buf_size = 1 << av_log2(s->win_size);
s->pts = AV_NOPTS_VALUE;
s->window_func_lut = av_calloc(s->buf_size, sizeof(*s->window_func_lut));
if (!s->window_func_lut)
@ -1540,16 +1535,21 @@ fail:
static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioSurroundContext *s = ctx->priv;
float *src = (float *)s->input_in->extended_data[ch];
float *win = (float *)s->window->extended_data[ch];
const int offset = s->buf_size - s->hop_size;
const float level_in = s->input_levels[ch];
float *dst;
int n;
AVFrame *in = arg;
dst = (float *)s->input_in->extended_data[ch];
for (n = 0; n < s->buf_size; n++) {
dst[n] *= s->window_func_lut[n] * level_in;
memmove(src, &src[s->hop_size], offset * sizeof(float));
memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
memset(&src[offset + in->nb_samples], 0, (s->hop_size - in->nb_samples) * sizeof(float));
for (int n = 0; n < s->buf_size; n++) {
win[n] = src[n] * s->window_func_lut[n] * level_in;
}
s->tx_fn(s->rdft[ch], (float *)s->input->extended_data[ch], dst, sizeof(float));
s->tx_fn(s->rdft[ch], (float *)s->input->extended_data[ch], win, sizeof(float));
return 0;
}
@ -1583,19 +1583,14 @@ static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
return 0;
}
static int filter_frame(AVFilterLink *inlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
AVFrame *out;
int ret;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input_in->extended_data, s->buf_size);
if (ret < 0)
return ret;
ff_filter_execute(ctx, fft_channel, NULL, NULL, inlink->channels);
ff_filter_execute(ctx, fft_channel, in, NULL, inlink->channels);
s->filter(ctx);
@ -1605,11 +1600,10 @@ static int filter_frame(AVFilterLink *inlink)
ff_filter_execute(ctx, ifft_channel, out, NULL, outlink->channels);
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_audio_fifo_drain(s->fifo, FFMIN(av_audio_fifo_size(s->fifo), s->hop_size));
out->pts = in->pts;
out->nb_samples = in->nb_samples;
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
@ -1624,48 +1618,21 @@ static int activate(AVFilterContext *ctx)
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (!s->eof && av_audio_fifo_size(s->fifo) < s->buf_size) {
ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
if (ret > 0) {
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;
}
}
if ((av_audio_fifo_size(s->fifo) >= s->buf_size) ||
(av_audio_fifo_size(s->fifo) > 0 && s->eof)) {
ret = filter_frame(inlink);
if (av_audio_fifo_size(s->fifo) >= s->buf_size)
ff_filter_set_ready(ctx, 100);
ret = ff_inlink_consume_samples(inlink, s->hop_size, s->hop_size, &in);
if (ret < 0)
return ret;
}
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF) {
s->eof = 1;
if (av_audio_fifo_size(s->fifo) >= 0) {
ff_filter_set_ready(ctx, 100);
return 0;
}
}
}
if (ret > 0)
ret = filter_frame(inlink, in);
if (ret < 0)
return ret;
if (s->eof && av_audio_fifo_size(s->fifo) <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
}
if (!s->eof)
FF_FILTER_FORWARD_WANTED(outlink, inlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
@ -1674,6 +1641,7 @@ static av_cold void uninit(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
av_frame_free(&s->window);
av_frame_free(&s->input_in);
av_frame_free(&s->input);
av_frame_free(&s->output);
@ -1688,7 +1656,6 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->output_levels);
av_freep(&s->rdft);
av_freep(&s->irdft);
av_audio_fifo_free(s->fifo);
av_freep(&s->window_func_lut);
}