From fee804f7edcc974507de9a84cb1d0ba702c8935b Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Tue, 22 Feb 2022 13:02:41 +0100 Subject: avfilter/af_surround: do not rewrite pts any more Also stop using fifo and excessive peeking. --- libavfilter/af_surround.c | 97 ++++++++++++++++------------------------------- 1 file changed, 32 insertions(+), 65 deletions(-) diff --git a/libavfilter/af_surround.c b/libavfilter/af_surround.c index c5657e405d..2a1208f703 100644 --- a/libavfilter/af_surround.c +++ b/libavfilter/af_surround.c @@ -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); } -- cgit v1.2.3