mirror of https://git.videolan.org/git/ffmpeg.git
avfilter/af_dynaudnorm: reduce number of lines by using for (int ...
This commit is contained in:
Paul B Mahol
parent
aa6b9066b9
commit
178d8036dc
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@ -248,7 +248,6 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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double total_weight = 0.0;
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const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
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double adjust;
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int i;
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// Pre-compute constants
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const int offset = s->filter_size / 2;
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@ -256,7 +255,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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const double c2 = 2.0 * sigma * sigma;
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// Compute weights
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for (i = 0; i < s->filter_size; i++) {
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for (int i = 0; i < s->filter_size; i++) {
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const int x = i - offset;
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s->weights[i] = c1 * exp(-x * x / c2);
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@ -265,7 +264,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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// Adjust weights
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adjust = 1.0 / total_weight;
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for (i = 0; i < s->filter_size; i++) {
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for (int i = 0; i < s->filter_size; i++) {
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s->weights[i] *= adjust;
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}
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}
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@ -273,13 +272,12 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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static av_cold void uninit(AVFilterContext *ctx)
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{
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DynamicAudioNormalizerContext *s = ctx->priv;
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int c;
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av_freep(&s->prev_amplification_factor);
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av_freep(&s->dc_correction_value);
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av_freep(&s->compress_threshold);
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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if (s->gain_history_original)
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cqueue_free(s->gain_history_original[c]);
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if (s->gain_history_minimum)
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@ -309,7 +307,6 @@ static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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DynamicAudioNormalizerContext *s = ctx->priv;
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int c;
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uninit(ctx);
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@ -333,7 +330,7 @@ static int config_input(AVFilterLink *inlink)
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!s->is_enabled || !s->weights)
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return AVERROR(ENOMEM);
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for (c = 0; c < inlink->channels; c++) {
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for (int c = 0; c < inlink->channels; c++) {
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s->prev_amplification_factor[c] = 1.0;
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s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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@ -378,19 +375,18 @@ static inline double bound(const double threshold, const double val)
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static double find_peak_magnitude(AVFrame *frame, int channel)
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{
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double max = DBL_EPSILON;
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int c, i;
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if (channel == -1) {
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for (c = 0; c < frame->channels; c++) {
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for (int c = 0; c < frame->channels; c++) {
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double *data_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++)
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for (int i = 0; i < frame->nb_samples; i++)
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max = fmax(max, fabs(data_ptr[i]));
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}
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} else {
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double *data_ptr = (double *)frame->extended_data[channel];
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for (i = 0; i < frame->nb_samples; i++)
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for (int i = 0; i < frame->nb_samples; i++)
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max = fmax(max, fabs(data_ptr[i]));
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}
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@ -400,13 +396,12 @@ static double find_peak_magnitude(AVFrame *frame, int channel)
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static double compute_frame_rms(AVFrame *frame, int channel)
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{
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double rms_value = 0.0;
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int c, i;
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if (channel == -1) {
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for (c = 0; c < frame->channels; c++) {
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for (int c = 0; c < frame->channels; c++) {
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const double *data_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++) {
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for (int i = 0; i < frame->nb_samples; i++) {
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rms_value += pow_2(data_ptr[i]);
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}
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}
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@ -414,7 +409,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
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rms_value /= frame->nb_samples * frame->channels;
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} else {
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const double *data_ptr = (double *)frame->extended_data[channel];
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for (i = 0; i < frame->nb_samples; i++) {
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for (int i = 0; i < frame->nb_samples; i++) {
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rms_value += pow_2(data_ptr[i]);
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}
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@ -441,9 +436,8 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *
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static double minimum_filter(cqueue *q)
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{
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double min = DBL_MAX;
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int i;
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for (i = 0; i < cqueue_size(q); i++) {
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for (int i = 0; i < cqueue_size(q); i++) {
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min = fmin(min, cqueue_peek(q, i));
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}
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@ -454,9 +448,8 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu
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{
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const double *weights = s->weights;
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double result = 0.0, tsum = 0.0;
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int i;
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for (i = 0; i < cqueue_size(q); i++) {
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for (int i = 0; i < cqueue_size(q); i++) {
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double tq_item = cqueue_peek(tq, i);
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double q_item = cqueue_peek(q, i);
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@ -540,21 +533,20 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra
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{
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const double diff = 1.0 / frame->nb_samples;
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int is_first_frame = cqueue_empty(s->gain_history_original[0]);
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int c, i;
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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const int bypass = bypass_channel(s, frame, c);
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double *dst_ptr = (double *)frame->extended_data[c];
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double current_average_value = 0.0;
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double prev_value;
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for (i = 0; i < frame->nb_samples; i++)
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for (int i = 0; i < frame->nb_samples; i++)
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current_average_value += dst_ptr[i] * diff;
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prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
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s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
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for (i = 0; i < frame->nb_samples && !bypass; i++) {
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for (int i = 0; i < frame->nb_samples && !bypass; i++) {
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dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
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}
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}
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@ -586,13 +578,12 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
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AVFrame *frame, int channel)
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{
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double variance = 0.0;
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int i, c;
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if (channel == -1) {
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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const double *data_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++) {
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for (int i = 0; i < frame->nb_samples; i++) {
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variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
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}
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}
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@ -600,7 +591,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
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} else {
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const double *data_ptr = (double *)frame->extended_data[channel];
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for (i = 0; i < frame->nb_samples; i++) {
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for (int i = 0; i < frame->nb_samples; i++) {
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variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
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}
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variance /= frame->nb_samples - 1;
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@ -612,7 +603,6 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
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static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
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{
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int is_first_frame = cqueue_empty(s->gain_history_original[0]);
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int c, i;
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if (s->channels_coupled) {
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const double standard_deviation = compute_frame_std_dev(s, frame, -1);
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@ -625,20 +615,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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prev_actual_thresh = setup_compress_thresh(prev_value);
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curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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double *const dst_ptr = (double *)frame->extended_data[c];
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const int bypass = bypass_channel(s, frame, c);
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if (bypass)
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continue;
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for (i = 0; i < frame->nb_samples; i++) {
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for (int i = 0; i < frame->nb_samples; i++) {
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
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dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
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}
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}
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} else {
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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const int bypass = bypass_channel(s, frame, c);
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const double standard_deviation = compute_frame_std_dev(s, frame, c);
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const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation));
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@ -652,7 +642,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
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dst_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples && !bypass; i++) {
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for (int i = 0; i < frame->nb_samples && !bypass; i++) {
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const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
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dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
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}
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@ -718,14 +708,10 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
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if (s->channels_coupled) {
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const local_gain gain = get_max_local_gain(s, analyze_frame, -1);
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int c;
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for (c = 0; c < s->channels; c++)
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for (int c = 0; c < s->channels; c++)
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update_gain_history(s, c, gain);
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} else {
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int c;
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for (c = 0; c < s->channels; c++)
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for (int c = 0; c < s->channels; c++)
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update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
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}
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@ -735,9 +721,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
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static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in,
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AVFrame *frame, int enabled)
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{
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int c, i;
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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const int bypass = bypass_channel(s, frame, c);
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const double *src_ptr = (const double *)in->extended_data[c];
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double *dst_ptr = (double *)frame->extended_data[c];
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@ -745,7 +729,7 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in,
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cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor);
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for (i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
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for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
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const double amplification_factor = fade(s->prev_amplification_factor[c],
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current_amplification_factor, i,
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frame->nb_samples);
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@ -811,15 +795,14 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
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AVFilterLink *outlink)
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{
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AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance);
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int c, i;
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if (!out)
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return AVERROR(ENOMEM);
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for (c = 0; c < s->channels; c++) {
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for (int c = 0; c < s->channels; c++) {
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double *dst_ptr = (double *)out->extended_data[c];
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for (i = 0; i < out->nb_samples; i++) {
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for (int i = 0; i < out->nb_samples; i++) {
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dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value);
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if (s->dc_correction) {
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dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
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