ffmpeg | branch: master | Paul B Mahol <one...@gmail.com> | Wed Feb 23 10:20:58 2022 +0100| [57f0cdbe17dfe5b304898aa6d05ab9df4bdb284d] | committer: Paul B Mahol
avfilter/af_loudnorm: fix filtering of last 2.9 seconds > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=57f0cdbe17dfe5b304898aa6d05ab9df4bdb284d --- libavfilter/af_loudnorm.c | 131 +++++++++++++--------------------------------- 1 file changed, 36 insertions(+), 95 deletions(-) diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c index 9bb0c65bb7..493306c707 100644 --- a/libavfilter/af_loudnorm.c +++ b/libavfilter/af_loudnorm.c @@ -408,37 +408,45 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFilterContext *ctx = inlink->dst; LoudNormContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; + const int final_samples = FFMIN(19200, inlink->sample_count_out - outlink->sample_count_in); AVFrame *out; - const double *src; + const double *src = NULL; double *dst; double *buf; double *limiter_buf; - int i, n, c, subframe_length, src_index; + int n, c, subframe_length; double gain, gain_next, env_global, env_shortterm, global, shortterm, lra, relative_threshold; - if (av_frame_is_writable(in)) { - out = in; - } else { - out = ff_get_audio_buffer(outlink, in->nb_samples); + if (in) { + if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) + s->frame_type = LINEAR_MODE; + + out = ff_get_audio_buffer(outlink, s->frame_type == LINEAR_MODE ? in->nb_samples : 19200); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); + } else { + out = ff_get_audio_buffer(outlink, 19200); + if (!out) + return AVERROR(ENOMEM); } out->pts = s->pts[0]; memmove(s->pts, &s->pts[1], (FF_ARRAY_ELEMS(s->pts) - 1) * sizeof(s->pts[0])); - src = (const double *)in->data[0]; dst = (double *)out->data[0]; buf = s->buf; limiter_buf = s->limiter_buf; - ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples); + if (in) { + src = (const double *)in->data[0]; + ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples); + } - if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) { + if (s->frame_type == FIRST_FRAME && in && in->nb_samples < frame_size(inlink->sample_rate, 3000)) { double offset, offset_tp, true_peak; ff_ebur128_loudness_global(s->r128_in, &global); @@ -452,7 +460,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) offset = pow(10., (s->target_i - global) / 20.); offset_tp = true_peak * offset; s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak; - s->frame_type = LINEAR_MODE; } switch (s->frame_type) { @@ -502,16 +509,19 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) s->frame_type = INNER_FRAME; break; + case FINAL_FRAME: case INNER_FRAME: gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30); gain_next = gaussian_filter(s, s->index + 11 < 30 ? s->index + 11 : s->index + 11 - 30); - for (n = 0; n < in->nb_samples; n++) { + for (n = 0; n < out->nb_samples; n++) { for (c = 0; c < inlink->channels; c++) { - buf[s->prev_buf_index + c] = src[c]; - limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / in->nb_samples) * (gain_next - gain))) * s->offset; + if (src) + buf[s->prev_buf_index + c] = src[c]; + limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / out->nb_samples) * (gain_next - gain))) * s->offset; } - src += inlink->channels; + if (src) + src += inlink->channels; s->limiter_buf_index += inlink->channels; if (s->limiter_buf_index >= s->limiter_buf_size) @@ -526,11 +536,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) s->buf_index -= s->buf_size; } - subframe_length = (frame_size(inlink->sample_rate, 100) - in->nb_samples) * inlink->channels; + subframe_length = (frame_size(inlink->sample_rate, 100) - out->nb_samples) * inlink->channels; s->limiter_buf_index = s->limiter_buf_index + subframe_length < s->limiter_buf_size ? s->limiter_buf_index + subframe_length : s->limiter_buf_index + subframe_length - s->limiter_buf_size; - true_peak_limiter(s, dst, in->nb_samples, inlink->channels); - ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); + true_peak_limiter(s, dst, out->nb_samples, inlink->channels); + ff_ebur128_add_frames_double(s->r128_out, dst, out->nb_samples); ff_ebur128_loudness_range(s->r128_in, &lra); ff_ebur128_loudness_global(s->r128_in, &global); @@ -560,51 +570,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) s->index++; if (s->index >= 30) s->index -= 30; - s->prev_nb_samples = in->nb_samples; - break; - - case FINAL_FRAME: - gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30); - s->limiter_buf_index = 0; - src_index = 0; - - for (n = 0; n < s->limiter_buf_size / inlink->channels; n++) { - for (c = 0; c < inlink->channels; c++) { - s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset; - } - src_index += inlink->channels; - - s->limiter_buf_index += inlink->channels; - if (s->limiter_buf_index >= s->limiter_buf_size) - s->limiter_buf_index -= s->limiter_buf_size; - } - - subframe_length = frame_size(inlink->sample_rate, 100); - for (i = 0; i < in->nb_samples / subframe_length; i++) { - true_peak_limiter(s, dst, subframe_length, inlink->channels); - - for (n = 0; n < subframe_length; n++) { - for (c = 0; c < inlink->channels; c++) { - if (src_index < (in->nb_samples * inlink->channels)) { - limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset; - } else { - limiter_buf[s->limiter_buf_index + c] = 0.; - } - } - - if (src_index < (in->nb_samples * inlink->channels)) - src_index += inlink->channels; - - s->limiter_buf_index += inlink->channels; - if (s->limiter_buf_index >= s->limiter_buf_size) - s->limiter_buf_index -= s->limiter_buf_size; - } - - dst += (subframe_length * inlink->channels); - } - - dst = (double *)out->data[0]; - ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); + if (s->frame_type == FINAL_FRAME) + out->nb_samples = final_samples; + s->prev_nb_samples = out->nb_samples; break; case LINEAR_MODE: @@ -617,11 +585,12 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) } dst = (double *)out->data[0]; + out->nb_samples = in->nb_samples; ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); break; } - if (in != out) + if (in) av_frame_free(&in); return ff_filter_frame(outlink, out); } @@ -634,38 +603,9 @@ static int flush_frame(AVFilterLink *outlink) int ret = 0; if (s->frame_type == INNER_FRAME) { - double *src; - double *buf; - int nb_samples, n, c, offset; - AVFrame *frame; - - nb_samples = (s->buf_size / inlink->channels) - s->prev_nb_samples; - nb_samples -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples); - - frame = ff_get_audio_buffer(outlink, nb_samples); - if (!frame) - return AVERROR(ENOMEM); - frame->nb_samples = nb_samples; - - buf = s->buf; - src = (double *)frame->data[0]; - - offset = ((s->limiter_buf_size / inlink->channels) - s->prev_nb_samples) * inlink->channels; - offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->channels; - s->buf_index = s->buf_index - offset < 0 ? s->buf_index - offset + s->buf_size : s->buf_index - offset; - - for (n = 0; n < nb_samples; n++) { - for (c = 0; c < inlink->channels; c++) { - src[c] = buf[s->buf_index + c]; - } - src += inlink->channels; - s->buf_index += inlink->channels; - if (s->buf_index >= s->buf_size) - s->buf_index -= s->buf_size; - } - s->frame_type = FINAL_FRAME; - ret = filter_frame(inlink, frame); + while (inlink->sample_count_out > outlink->sample_count_in) + ret = filter_frame(inlink, NULL); } return ret; } @@ -712,8 +652,9 @@ static int activate(AVFilterContext *ctx) return ret; if (ff_inlink_acknowledge_status(inlink, &status, &pts)) { + ret = flush_frame(outlink); ff_outlink_set_status(outlink, status, pts); - return flush_frame(outlink); + return ret; } FF_FILTER_FORWARD_WANTED(outlink, inlink); _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email ffmpeg-cvslog-requ...@ffmpeg.org with subject "unsubscribe".
