From: Niklas Haas <g...@haasn.dev> This is actually allowed by non-ancient versions of C. --- libavfilter/f_ebur128.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-)
diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c index 4051b1ea95..1fb7129271 100644 --- a/libavfilter/f_ebur128.c +++ b/libavfilter/f_ebur128.c @@ -652,7 +652,7 @@ void ff_ebur128_filter_channels_c(const EBUR128DSPContext *dsp, static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) { - int i, ch, idx_insample, ret; + int ret; AVFilterContext *ctx = inlink->dst; EBUR128Context *ebur128 = ctx->priv; const EBUR128DSPContext *dsp = &ebur128->dsp; @@ -705,7 +705,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) } - for (idx_insample = ebur128->idx_insample; idx_insample < nb_samples; idx_insample++) { + for (int idx_insample = ebur128->idx_insample; idx_insample < nb_samples; idx_insample++) { const int bin_id_400 = ebur128->i400.cache_pos++; const int bin_id_3000 = ebur128->i3000.cache_pos++; @@ -741,7 +741,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) #define COMPUTE_LOUDNESS(m, time) do { \ if (ebur128->i##time.filled) { \ /* weighting sum of the last <time> ms */ \ - for (ch = 0; ch < nb_channels; ch++) \ + for (int ch = 0; ch < nb_channels; ch++) \ power_##time += ebur128->ch_weighting[ch] * ebur128->i##time.sum[ch]; \ power_##time /= I##time##_BINS(inlink->sample_rate); \ } \ @@ -762,7 +762,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) /* compute integrated loudness by summing the histogram values * above the relative threshold */ - for (i = gate_hist_pos; i < HIST_SIZE; i++) { + for (int i = gate_hist_pos; i < HIST_SIZE; i++) { const unsigned nb_v = ebur128->i400.histogram[i].count; nb_integrated += nb_v; integrated_sum += nb_v * ebur128->i400.histogram[i].energy; @@ -788,7 +788,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) int gate_hist_pos = gate_update(&ebur128->i3000, power_3000, loudness_3000, LRA_GATE_THRES); - for (i = gate_hist_pos; i < HIST_SIZE; i++) + for (int i = gate_hist_pos; i < HIST_SIZE; i++) nb_powers += ebur128->i3000.histogram[i].count; if (nb_powers) { uint64_t n, nb_pow; @@ -796,7 +796,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) /* get lower loudness to consider */ n = 0; nb_pow = LRA_LOWER_PRC * nb_powers * 0.01 + 0.5; - for (i = gate_hist_pos; i < HIST_SIZE; i++) { + for (int i = gate_hist_pos; i < HIST_SIZE; i++) { n += ebur128->i3000.histogram[i].count; if (n >= nb_pow) { ebur128->lra_low = ebur128->i3000.histogram[i].loudness; @@ -807,7 +807,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) /* get higher loudness to consider */ n = nb_powers; nb_pow = LRA_HIGHER_PRC * nb_powers * 0.01 + 0.5; - for (i = HIST_SIZE - 1; i >= 0; i--) { + for (int i = HIST_SIZE - 1; i >= 0; i--) { n -= FFMIN(n, ebur128->i3000.histogram[i].count); if (n < nb_pow) { ebur128->lra_high = ebur128->i3000.histogram[i].loudness; @@ -909,7 +909,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) if (ebur128->peak_mode & PEAK_MODE_ ## ptype ## _PEAKS) { \ double max_peak = 0.0; \ char key[64]; \ - for (ch = 0; ch < nb_channels; ch++) { \ + for (int ch = 0; ch < nb_channels; ch++) { \ snprintf(key, sizeof(key), \ META_PREFIX AV_STRINGIFY(name) "_peaks_ch%d", ch); \ max_peak = fmax(max_peak, ebur128->name##_peaks[ch]); \ @@ -948,7 +948,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) #define PRINT_PEAKS(str, sp, ptype) do { \ if (ebur128->peak_mode & PEAK_MODE_ ## ptype ## _PEAKS) { \ av_log(ctx, ebur128->loglevel, " " str ":"); \ - for (ch = 0; ch < nb_channels; ch++) \ + for (int ch = 0; ch < nb_channels; ch++) \ av_log(ctx, ebur128->loglevel, " %5.1f", DBFS(sp[ch])); \ av_log(ctx, ebur128->loglevel, " dBFS"); \ } \ -- 2.49.0 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".