ffmpeg | branch: master | Paul B Mahol <onemda@gmail.com> | Mon Feb 28 21:54:28 2022 +0100| [178d8036dc4bb5f20b58e6a02da3d596b2f6f615] | committer: Paul B Mahol avfilter/af_dynaudnorm: reduce number of lines by using for (int ...

http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=178d8036dc4bb5f20b...

libavfilter/af_dynaudnorm.c | 73 +++++++++++++++++---------------------------- 1 file changed, 28 insertions(+), 45 deletions(-) diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c index 19f3b528d9..837126ac0d 100644 --- a/libavfilter/af_dynaudnorm.c +++ b/libavfilter/af_dynaudnorm.c @@ -248,7 +248,6 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) double total_weight = 0.0; const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0); double adjust; - int i; // Pre-compute constants const int offset = s->filter_size / 2; @@ -256,7 +255,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) const double c2 = 2.0 * sigma * sigma; // Compute weights - for (i = 0; i < s->filter_size; i++) { + for (int i = 0; i < s->filter_size; i++) { const int x = i - offset; s->weights[i] = c1 * exp(-x * x / c2); @@ -265,7 +264,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) // Adjust weights adjust = 1.0 / total_weight; - for (i = 0; i < s->filter_size; i++) { + for (int i = 0; i < s->filter_size; i++) { s->weights[i] *= adjust; } } @@ -273,13 +272,12 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s) static av_cold void uninit(AVFilterContext *ctx) { DynamicAudioNormalizerContext *s = ctx->priv; - int c; av_freep(&s->prev_amplification_factor); av_freep(&s->dc_correction_value); av_freep(&s->compress_threshold); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { if (s->gain_history_original) cqueue_free(s->gain_history_original[c]); if (s->gain_history_minimum) @@ -309,7 +307,6 @@ static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; DynamicAudioNormalizerContext *s = ctx->priv; - int c; uninit(ctx); @@ -333,7 +330,7 @@ static int config_input(AVFilterLink *inlink) !s->is_enabled || !s->weights) return AVERROR(ENOMEM); - for (c = 0; c < inlink->channels; c++) { + for (int c = 0; c < inlink->channels; c++) { s->prev_amplification_factor[c] = 1.0; s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); @@ -378,19 +375,18 @@ static inline double bound(const double threshold, const double val) static double find_peak_magnitude(AVFrame *frame, int channel) { double max = DBL_EPSILON; - int c, i; if (channel == -1) { - for (c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->channels; c++) { double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } } else { double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) max = fmax(max, fabs(data_ptr[i])); } @@ -400,13 +396,12 @@ static double find_peak_magnitude(AVFrame *frame, int channel) static double compute_frame_rms(AVFrame *frame, int channel) { double rms_value = 0.0; - int c, i; if (channel == -1) { - for (c = 0; c < frame->channels; c++) { + for (int c = 0; c < frame->channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } } @@ -414,7 +409,7 @@ static double compute_frame_rms(AVFrame *frame, int channel) rms_value /= frame->nb_samples * frame->channels; } else { const double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { rms_value += pow_2(data_ptr[i]); } @@ -441,9 +436,8 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame * static double minimum_filter(cqueue *q) { double min = DBL_MAX; - int i; - for (i = 0; i < cqueue_size(q); i++) { + for (int i = 0; i < cqueue_size(q); i++) { min = fmin(min, cqueue_peek(q, i)); } @@ -454,9 +448,8 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu { const double *weights = s->weights; double result = 0.0, tsum = 0.0; - int i; - for (i = 0; i < cqueue_size(q); i++) { + for (int i = 0; i < cqueue_size(q); i++) { double tq_item = cqueue_peek(tq, i); double q_item = cqueue_peek(q, i); @@ -540,21 +533,20 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra { const double diff = 1.0 / frame->nb_samples; int is_first_frame = cqueue_empty(s->gain_history_original[0]); - int c, i; - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); double *dst_ptr = (double *)frame->extended_data[c]; double current_average_value = 0.0; double prev_value; - for (i = 0; i < frame->nb_samples; i++) + for (int i = 0; i < frame->nb_samples; i++) current_average_value += dst_ptr[i] * diff; prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c]; s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1); - for (i = 0; i < frame->nb_samples && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && !bypass; i++) { dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples); } } @@ -586,13 +578,12 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, AVFrame *frame, int channel) { double variance = 0.0; - int i, c; if (channel == -1) { - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const double *data_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* } } @@ -600,7 +591,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, } else { const double *data_ptr = (double *)frame->extended_data[channel]; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* } variance /= frame->nb_samples - 1; @@ -612,7 +603,6 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame) { int is_first_frame = cqueue_empty(s->gain_history_original[0]); - int c, i; if (s->channels_coupled) { const double standard_deviation = compute_frame_std_dev(s, frame, -1); @@ -625,20 +615,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame prev_actual_thresh = setup_compress_thresh(prev_value); curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { double *const dst_ptr = (double *)frame->extended_data[c]; const int bypass = bypass_channel(s, frame, c); if (bypass) continue; - for (i = 0; i < frame->nb_samples; i++) { + for (int i = 0; i < frame->nb_samples; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } } } else { - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double standard_deviation = compute_frame_std_dev(s, frame, c); const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation)); @@ -652,7 +642,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]); dst_ptr = (double *)frame->extended_data[c]; - for (i = 0; i < frame->nb_samples && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && !bypass; i++) { const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); } @@ -718,14 +708,10 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink if (s->channels_coupled) { const local_gain gain = get_max_local_gain(s, analyze_frame, -1); - int c; - - for (c = 0; c < s->channels; c++) + for (int c = 0; c < s->channels; c++) update_gain_history(s, c, gain); } else { - int c; - - for (c = 0; c < s->channels; c++) + for (int c = 0; c < s->channels; c++) update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c)); } @@ -735,9 +721,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, AVFrame *frame, int enabled) { - int c, i; - - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { const int bypass = bypass_channel(s, frame, c); const double *src_ptr = (const double *)in->extended_data[c]; double *dst_ptr = (double *)frame->extended_data[c]; @@ -745,7 +729,7 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in, cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor); - for (i = 0; i < frame->nb_samples && enabled && !bypass; i++) { + for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) { const double amplification_factor = fade(s->prev_amplification_factor[c], current_amplification_factor, i, frame->nb_samples); @@ -811,15 +795,14 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, AVFilterLink *outlink) { AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance); - int c, i; if (!out) return AVERROR(ENOMEM); - for (c = 0; c < s->channels; c++) { + for (int c = 0; c < s->channels; c++) { double *dst_ptr = (double *)out->extended_data[c]; - for (i = 0; i < out->nb_samples; i++) { + for (int i = 0; i < out->nb_samples; i++) { dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value); if (s->dc_correction) { dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;