filter_charcoal.o \
filter_dynamictext.o \
filter_invert.o \
+ filter_loudness.o \
filter_sepia.o \
producer_blipflash.o \
producer_count.o \
- transition_affine.o
+ transition_affine.o \
+ ebur128/ebur128.o
SRCS := $(OBJS:.o=.c)
--- /dev/null
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--- /dev/null
+/* See COPYING file for copyright and license details. */
+
+#include "ebur128.h"
+
+#include <float.h>
+#include <limits.h>
+#include <math.h> /* You may have to define _USE_MATH_DEFINES if you use MSVC */
+#include <stdio.h>
+#include <stdlib.h>
+
+/* This can be replaced by any BSD-like queue implementation. */
+#include <sys/queue.h>
+
+#ifdef USE_SPEEX_RESAMPLER
+ #include <speex/speex_resampler.h>
+#endif
+
+#define CHECK_ERROR(condition, errorcode, goto_point) \
+ if ((condition)) { \
+ errcode = (errorcode); \
+ goto goto_point; \
+ }
+
+SLIST_HEAD(ebur128_double_queue, ebur128_dq_entry);
+struct ebur128_dq_entry {
+ double z;
+ SLIST_ENTRY(ebur128_dq_entry) entries;
+};
+
+struct ebur128_state_internal {
+ /** Filtered audio data (used as ring buffer). */
+ double* audio_data;
+ /** Size of audio_data array. */
+ size_t audio_data_frames;
+ /** Current index for audio_data. */
+ size_t audio_data_index;
+ /** How many frames are needed for a gating block. Will correspond to 400ms
+ * of audio at initialization, and 100ms after the first block (75% overlap
+ * as specified in the 2011 revision of BS1770). */
+ unsigned long needed_frames;
+ /** The channel map. Has as many elements as there are channels. */
+ int* channel_map;
+ /** How many samples fit in 100ms (rounded). */
+ unsigned long samples_in_100ms;
+ /** BS.1770 filter coefficients (nominator). */
+ double b[5];
+ /** BS.1770 filter coefficients (denominator). */
+ double a[5];
+ /** BS.1770 filter state. */
+ double v[5][5];
+ /** Linked list of block energies. */
+ struct ebur128_double_queue block_list;
+ /** Linked list of 3s-block energies, used to calculate LRA. */
+ struct ebur128_double_queue short_term_block_list;
+ int use_histogram;
+ unsigned long *block_energy_histogram;
+ unsigned long *short_term_block_energy_histogram;
+ /** Keeps track of when a new short term block is needed. */
+ size_t short_term_frame_counter;
+ /** Maximum sample peak, one per channel */
+ double* sample_peak;
+ /** Maximum true peak, one per channel */
+ double* true_peak;
+#ifdef USE_SPEEX_RESAMPLER
+ SpeexResamplerState* resampler;
+#endif
+ size_t oversample_factor;
+ float* resampler_buffer_input;
+ size_t resampler_buffer_input_frames;
+ float* resampler_buffer_output;
+ size_t resampler_buffer_output_frames;
+};
+
+static double relative_gate = -10.0;
+
+/* Those will be calculated when initializing the library */
+static double relative_gate_factor;
+static double minus_twenty_decibels;
+static double histogram_energies[1000];
+static double histogram_energy_boundaries[1001];
+
+static void ebur128_init_filter(ebur128_state* st) {
+ int i, j;
+
+ double f0 = 1681.974450955533;
+ double G = 3.999843853973347;
+ double Q = 0.7071752369554196;
+
+ double K = tan(M_PI * f0 / (double) st->samplerate);
+ double Vh = pow(10.0, G / 20.0);
+ double Vb = pow(Vh, 0.4996667741545416);
+
+ double pb[3] = {0.0, 0.0, 0.0};
+ double pa[3] = {1.0, 0.0, 0.0};
+ double rb[3] = {1.0, -2.0, 1.0};
+ double ra[3] = {1.0, 0.0, 0.0};
+
+ double a0 = 1.0 + K / Q + K * K ;
+ pb[0] = (Vh + Vb * K / Q + K * K) / a0;
+ pb[1] = 2.0 * (K * K - Vh) / a0;
+ pb[2] = (Vh - Vb * K / Q + K * K) / a0;
+ pa[1] = 2.0 * (K * K - 1.0) / a0;
+ pa[2] = (1.0 - K / Q + K * K) / a0;
+
+ /* fprintf(stderr, "%.14f %.14f %.14f %.14f %.14f\n",
+ b1[0], b1[1], b1[2], a1[1], a1[2]); */
+
+ f0 = 38.13547087602444;
+ Q = 0.5003270373238773;
+ K = tan(M_PI * f0 / (double) st->samplerate);
+
+ ra[1] = 2.0 * (K * K - 1.0) / (1.0 + K / Q + K * K);
+ ra[2] = (1.0 - K / Q + K * K) / (1.0 + K / Q + K * K);
+
+ /* fprintf(stderr, "%.14f %.14f\n", a2[1], a2[2]); */
+
+ st->d->b[0] = pb[0] * rb[0];
+ st->d->b[1] = pb[0] * rb[1] + pb[1] * rb[0];
+ st->d->b[2] = pb[0] * rb[2] + pb[1] * rb[1] + pb[2] * rb[0];
+ st->d->b[3] = pb[1] * rb[2] + pb[2] * rb[1];
+ st->d->b[4] = pb[2] * rb[2];
+
+ st->d->a[0] = pa[0] * ra[0];
+ st->d->a[1] = pa[0] * ra[1] + pa[1] * ra[0];
+ st->d->a[2] = pa[0] * ra[2] + pa[1] * ra[1] + pa[2] * ra[0];
+ st->d->a[3] = pa[1] * ra[2] + pa[2] * ra[1];
+ st->d->a[4] = pa[2] * ra[2];
+
+ for (i = 0; i < 5; ++i) {
+ for (j = 0; j < 5; ++j) {
+ st->d->v[i][j] = 0.0;
+ }
+ }
+}
+
+static int ebur128_init_channel_map(ebur128_state* st) {
+ size_t i;
+ st->d->channel_map = (int*) malloc(st->channels * sizeof(int));
+ if (!st->d->channel_map) return EBUR128_ERROR_NOMEM;
+ if (st->channels == 4) {
+ st->d->channel_map[0] = EBUR128_LEFT;
+ st->d->channel_map[1] = EBUR128_RIGHT;
+ st->d->channel_map[2] = EBUR128_LEFT_SURROUND;
+ st->d->channel_map[3] = EBUR128_RIGHT_SURROUND;
+ } else if (st->channels == 5) {
+ st->d->channel_map[0] = EBUR128_LEFT;
+ st->d->channel_map[1] = EBUR128_RIGHT;
+ st->d->channel_map[2] = EBUR128_CENTER;
+ st->d->channel_map[3] = EBUR128_LEFT_SURROUND;
+ st->d->channel_map[4] = EBUR128_RIGHT_SURROUND;
+ } else {
+ for (i = 0; i < st->channels; ++i) {
+ switch (i) {
+ case 0: st->d->channel_map[i] = EBUR128_LEFT; break;
+ case 1: st->d->channel_map[i] = EBUR128_RIGHT; break;
+ case 2: st->d->channel_map[i] = EBUR128_CENTER; break;
+ case 3: st->d->channel_map[i] = EBUR128_UNUSED; break;
+ case 4: st->d->channel_map[i] = EBUR128_LEFT_SURROUND; break;
+ case 5: st->d->channel_map[i] = EBUR128_RIGHT_SURROUND; break;
+ default: st->d->channel_map[i] = EBUR128_UNUSED; break;
+ }
+ }
+ }
+ return EBUR128_SUCCESS;
+}
+
+#ifdef USE_SPEEX_RESAMPLER
+static int ebur128_init_resampler(ebur128_state* st) {
+ int errcode = EBUR128_SUCCESS;
+
+ if (st->samplerate < 96000) {
+ st->d->oversample_factor = 4;
+ } else if (st->samplerate < 192000) {
+ st->d->oversample_factor = 2;
+ } else {
+ st->d->oversample_factor = 1;
+ st->d->resampler_buffer_input = NULL;
+ st->d->resampler_buffer_output = NULL;
+ st->d->resampler = NULL;
+ }
+
+ st->d->resampler_buffer_input_frames = st->d->samples_in_100ms * 4;
+ st->d->resampler_buffer_input = malloc(st->d->resampler_buffer_input_frames *
+ st->channels *
+ sizeof(float));
+ CHECK_ERROR(!st->d->resampler_buffer_input, EBUR128_ERROR_NOMEM, exit)
+
+ st->d->resampler_buffer_output_frames =
+ st->d->resampler_buffer_input_frames *
+ st->d->oversample_factor;
+ st->d->resampler_buffer_output = malloc
+ (st->d->resampler_buffer_output_frames *
+ st->channels *
+ sizeof(float));
+ CHECK_ERROR(!st->d->resampler_buffer_output, EBUR128_ERROR_NOMEM, free_input)
+
+ st->d->resampler = speex_resampler_init
+ ((spx_uint32_t) st->channels,
+ (spx_uint32_t) st->samplerate,
+ (spx_uint32_t) (st->samplerate * st->d->oversample_factor),
+ 8, NULL);
+ CHECK_ERROR(!st->d->resampler, EBUR128_ERROR_NOMEM, free_output)
+
+ return errcode;
+
+free_output:
+ free(st->d->resampler_buffer_output);
+ st->d->resampler_buffer_output = NULL;
+free_input:
+ free(st->d->resampler_buffer_input);
+ st->d->resampler_buffer_input = NULL;
+exit:
+ return errcode;
+}
+
+static void ebur128_destroy_resampler(ebur128_state* st) {
+ free(st->d->resampler_buffer_input);
+ st->d->resampler_buffer_input = NULL;
+ free(st->d->resampler_buffer_output);
+ st->d->resampler_buffer_output = NULL;
+ speex_resampler_destroy(st->d->resampler);
+ st->d->resampler = NULL;
+}
+#endif
+
+void ebur128_get_version(int* major, int* minor, int* patch) {
+ *major = EBUR128_VERSION_MAJOR;
+ *minor = EBUR128_VERSION_MINOR;
+ *patch = EBUR128_VERSION_PATCH;
+}
+
+ebur128_state* ebur128_init(unsigned int channels,
+ unsigned long samplerate,
+ int mode) {
+ int errcode, result;
+ ebur128_state* st;
+ unsigned int i;
+
+ st = (ebur128_state*) malloc(sizeof(ebur128_state));
+ CHECK_ERROR(!st, 0, exit)
+ st->d = (struct ebur128_state_internal*)
+ malloc(sizeof(struct ebur128_state_internal));
+ CHECK_ERROR(!st->d, 0, free_state)
+ st->channels = channels;
+ errcode = ebur128_init_channel_map(st);
+ CHECK_ERROR(errcode, 0, free_internal)
+
+ st->d->sample_peak = (double*) malloc(channels * sizeof(double));
+ CHECK_ERROR(!st->d->sample_peak, 0, free_channel_map)
+ st->d->true_peak = (double*) malloc(channels * sizeof(double));
+ CHECK_ERROR(!st->d->true_peak, 0, free_sample_peak)
+ for (i = 0; i < channels; ++i) {
+ st->d->sample_peak[i] = 0.0;
+ st->d->true_peak[i] = 0.0;
+ }
+
+ st->d->use_histogram = mode & EBUR128_MODE_HISTOGRAM ? 1 : 0;
+
+ st->samplerate = samplerate;
+ st->d->samples_in_100ms = (st->samplerate + 5) / 10;
+ st->mode = mode;
+ if ((mode & EBUR128_MODE_S) == EBUR128_MODE_S) {
+ st->d->audio_data_frames = st->d->samples_in_100ms * 30;
+ } else if ((mode & EBUR128_MODE_M) == EBUR128_MODE_M) {
+ st->d->audio_data_frames = st->d->samples_in_100ms * 4;
+ } else {
+ return NULL;
+ }
+ st->d->audio_data = (double*) malloc(st->d->audio_data_frames *
+ st->channels *
+ sizeof(double));
+ CHECK_ERROR(!st->d->audio_data, 0, free_true_peak)
+ ebur128_init_filter(st);
+
+ if (st->d->use_histogram) {
+ st->d->block_energy_histogram = malloc(1000 * sizeof(unsigned long));
+ CHECK_ERROR(!st->d->block_energy_histogram, 0, free_audio_data)
+ for (i = 0; i < 1000; ++i) {
+ st->d->block_energy_histogram[i] = 0;
+ }
+ } else {
+ st->d->block_energy_histogram = NULL;
+ }
+ if (st->d->use_histogram) {
+ st->d->short_term_block_energy_histogram = malloc(1000 * sizeof(unsigned long));
+ CHECK_ERROR(!st->d->short_term_block_energy_histogram, 0, free_block_energy_histogram)
+ for (i = 0; i < 1000; ++i) {
+ st->d->short_term_block_energy_histogram[i] = 0;
+ }
+ } else {
+ st->d->short_term_block_energy_histogram = NULL;
+ }
+ SLIST_INIT(&st->d->block_list);
+ SLIST_INIT(&st->d->short_term_block_list);
+ st->d->short_term_frame_counter = 0;
+
+#ifdef USE_SPEEX_RESAMPLER
+ result = ebur128_init_resampler(st);
+ CHECK_ERROR(result, 0, free_short_term_block_energy_histogram)
+#endif
+
+ /* the first block needs 400ms of audio data */
+ st->d->needed_frames = st->d->samples_in_100ms * 4;
+ /* start at the beginning of the buffer */
+ st->d->audio_data_index = 0;
+
+ /* initialize static constants */
+ relative_gate_factor = pow(10.0, relative_gate / 10.0);
+ minus_twenty_decibels = pow(10.0, -20.0 / 10.0);
+ histogram_energy_boundaries[0] = pow(10.0, (-70.0 + 0.691) / 10.0);
+ if (st->d->use_histogram) {
+ for (i = 0; i < 1000; ++i) {
+ histogram_energies[i] = pow(10.0, ((double) i / 10.0 - 69.95 + 0.691) / 10.0);
+ }
+ for (i = 1; i < 1001; ++i) {
+ histogram_energy_boundaries[i] = pow(10.0, ((double) i / 10.0 - 70.0 + 0.691) / 10.0);
+ }
+ }
+
+ return st;
+
+free_short_term_block_energy_histogram:
+ free(st->d->short_term_block_energy_histogram);
+free_block_energy_histogram:
+ free(st->d->block_energy_histogram);
+free_audio_data:
+ free(st->d->audio_data);
+free_true_peak:
+ free(st->d->true_peak);
+free_sample_peak:
+ free(st->d->sample_peak);
+free_channel_map:
+ free(st->d->channel_map);
+free_internal:
+ free(st->d);
+free_state:
+ free(st);
+exit:
+ return NULL;
+}
+
+void ebur128_destroy(ebur128_state** st) {
+ struct ebur128_dq_entry* entry;
+ free((*st)->d->block_energy_histogram);
+ free((*st)->d->short_term_block_energy_histogram);
+ free((*st)->d->audio_data);
+ free((*st)->d->channel_map);
+ free((*st)->d->sample_peak);
+ free((*st)->d->true_peak);
+ while (!SLIST_EMPTY(&(*st)->d->block_list)) {
+ entry = SLIST_FIRST(&(*st)->d->block_list);
+ SLIST_REMOVE_HEAD(&(*st)->d->block_list, entries);
+ free(entry);
+ }
+ while (!SLIST_EMPTY(&(*st)->d->short_term_block_list)) {
+ entry = SLIST_FIRST(&(*st)->d->short_term_block_list);
+ SLIST_REMOVE_HEAD(&(*st)->d->short_term_block_list, entries);
+ free(entry);
+ }
+#ifdef USE_SPEEX_RESAMPLER
+ ebur128_destroy_resampler(*st);
+#endif
+
+ free((*st)->d);
+ free(*st);
+ *st = NULL;
+}
+
+static int ebur128_use_speex_resampler(ebur128_state* st) {
+#ifdef USE_SPEEX_RESAMPLER
+ return ((st->mode & EBUR128_MODE_TRUE_PEAK) == EBUR128_MODE_TRUE_PEAK);
+#else
+ (void) st;
+ return 0;
+#endif
+}
+
+static void ebur128_check_true_peak(ebur128_state* st, size_t frames) {
+#ifdef USE_SPEEX_RESAMPLER
+ size_t c, i;
+ spx_uint32_t in_len = (spx_uint32_t) frames;
+ spx_uint32_t out_len = (spx_uint32_t) st->d->resampler_buffer_output_frames;
+ speex_resampler_process_interleaved_float(
+ st->d->resampler,
+ st->d->resampler_buffer_input, &in_len,
+ st->d->resampler_buffer_output, &out_len);
+ for (c = 0; c < st->channels; ++c) {
+ for (i = 0; i < out_len; ++i) {
+ if (st->d->resampler_buffer_output[i * st->channels + c] >
+ st->d->true_peak[c]) {
+ st->d->true_peak[c] =
+ st->d->resampler_buffer_output[i * st->channels + c];
+ } else if (-st->d->resampler_buffer_output[i * st->channels + c] >
+ st->d->true_peak[c]) {
+ st->d->true_peak[c] =
+ -st->d->resampler_buffer_output[i * st->channels + c];
+ }
+ }
+ }
+#else
+ (void) st; (void) frames;
+#endif
+}
+
+#ifdef __SSE2_MATH__
+#include <xmmintrin.h>
+#define TURN_ON_FTZ \
+ unsigned int mxcsr = _mm_getcsr(); \
+ _mm_setcsr(mxcsr | _MM_FLUSH_ZERO_ON);
+#define TURN_OFF_FTZ _mm_setcsr(mxcsr);
+#define FLUSH_MANUALLY
+#else
+#warning "manual FTZ is being used, please enable SSE2 (-msse2 -mfpmath=sse)"
+#define TURN_ON_FTZ
+#define TURN_OFF_FTZ
+#define FLUSH_MANUALLY \
+ st->d->v[ci][4] = fabs(st->d->v[ci][4]) < DBL_MIN ? 0.0 : st->d->v[ci][4]; \
+ st->d->v[ci][3] = fabs(st->d->v[ci][3]) < DBL_MIN ? 0.0 : st->d->v[ci][3]; \
+ st->d->v[ci][2] = fabs(st->d->v[ci][2]) < DBL_MIN ? 0.0 : st->d->v[ci][2]; \
+ st->d->v[ci][1] = fabs(st->d->v[ci][1]) < DBL_MIN ? 0.0 : st->d->v[ci][1];
+#endif
+
+#define EBUR128_FILTER(type, min_scale, max_scale) \
+static void ebur128_filter_##type(ebur128_state* st, const type* src, \
+ size_t frames) { \
+ static double scaling_factor = -((double) min_scale) > (double) max_scale ? \
+ -((double) min_scale) : (double) max_scale; \
+ double* audio_data = st->d->audio_data + st->d->audio_data_index; \
+ size_t i, c; \
+ \
+ TURN_ON_FTZ \
+ \
+ if ((st->mode & EBUR128_MODE_SAMPLE_PEAK) == EBUR128_MODE_SAMPLE_PEAK) { \
+ for (c = 0; c < st->channels; ++c) { \
+ double max = 0.0; \
+ for (i = 0; i < frames; ++i) { \
+ if (src[i * st->channels + c] > max) { \
+ max = src[i * st->channels + c]; \
+ } else if (-src[i * st->channels + c] > max) { \
+ max = -1.0 * src[i * st->channels + c]; \
+ } \
+ } \
+ max /= scaling_factor; \
+ if (max > st->d->sample_peak[c]) st->d->sample_peak[c] = max; \
+ } \
+ } \
+ if (ebur128_use_speex_resampler(st)) { \
+ for (c = 0; c < st->channels; ++c) { \
+ for (i = 0; i < frames; ++i) { \
+ st->d->resampler_buffer_input[i * st->channels + c] = \
+ (float) (src[i * st->channels + c] / scaling_factor); \
+ } \
+ } \
+ ebur128_check_true_peak(st, frames); \
+ } \
+ for (c = 0; c < st->channels; ++c) { \
+ int ci = st->d->channel_map[c] - 1; \
+ if (ci < 0) continue; \
+ else if (ci > 4) ci = 0; /* dual mono */ \
+ for (i = 0; i < frames; ++i) { \
+ st->d->v[ci][0] = (double) (src[i * st->channels + c] / scaling_factor) \
+ - st->d->a[1] * st->d->v[ci][1] \
+ - st->d->a[2] * st->d->v[ci][2] \
+ - st->d->a[3] * st->d->v[ci][3] \
+ - st->d->a[4] * st->d->v[ci][4]; \
+ audio_data[i * st->channels + c] = \
+ st->d->b[0] * st->d->v[ci][0] \
+ + st->d->b[1] * st->d->v[ci][1] \
+ + st->d->b[2] * st->d->v[ci][2] \
+ + st->d->b[3] * st->d->v[ci][3] \
+ + st->d->b[4] * st->d->v[ci][4]; \
+ st->d->v[ci][4] = st->d->v[ci][3]; \
+ st->d->v[ci][3] = st->d->v[ci][2]; \
+ st->d->v[ci][2] = st->d->v[ci][1]; \
+ st->d->v[ci][1] = st->d->v[ci][0]; \
+ } \
+ FLUSH_MANUALLY \
+ } \
+ TURN_OFF_FTZ \
+}
+EBUR128_FILTER(short, SHRT_MIN, SHRT_MAX)
+EBUR128_FILTER(int, INT_MIN, INT_MAX)
+EBUR128_FILTER(float, -1.0f, 1.0f)
+EBUR128_FILTER(double, -1.0, 1.0)
+
+static double ebur128_energy_to_loudness(double energy) {
+ return 10 * (log(energy) / log(10.0)) - 0.691;
+}
+
+static size_t find_histogram_index(double energy) {
+ size_t index_min = 0;
+ size_t index_max = 1000;
+ size_t index_mid;
+
+ do {
+ index_mid = (index_min + index_max) / 2;
+ if (energy >= histogram_energy_boundaries[index_mid]) {
+ index_min = index_mid;
+ } else {
+ index_max = index_mid;
+ }
+ } while (index_max - index_min != 1);
+
+ return index_min;
+}
+
+static int ebur128_calc_gating_block(ebur128_state* st, size_t frames_per_block,
+ double* optional_output) {
+ size_t i, c;
+ double sum = 0.0;
+ double channel_sum;
+ for (c = 0; c < st->channels; ++c) {
+ if (st->d->channel_map[c] == EBUR128_UNUSED) continue;
+ channel_sum = 0.0;
+ if (st->d->audio_data_index < frames_per_block * st->channels) {
+ for (i = 0; i < st->d->audio_data_index / st->channels; ++i) {
+ channel_sum += st->d->audio_data[i * st->channels + c] *
+ st->d->audio_data[i * st->channels + c];
+ }
+ for (i = st->d->audio_data_frames -
+ (frames_per_block -
+ st->d->audio_data_index / st->channels);
+ i < st->d->audio_data_frames; ++i) {
+ channel_sum += st->d->audio_data[i * st->channels + c] *
+ st->d->audio_data[i * st->channels + c];
+ }
+ } else {
+ for (i = st->d->audio_data_index / st->channels - frames_per_block;
+ i < st->d->audio_data_index / st->channels;
+ ++i) {
+ channel_sum += st->d->audio_data[i * st->channels + c] *
+ st->d->audio_data[i * st->channels + c];
+ }
+ }
+ if (st->d->channel_map[c] == EBUR128_LEFT_SURROUND ||
+ st->d->channel_map[c] == EBUR128_RIGHT_SURROUND) {
+ channel_sum *= 1.41;
+ } else if (st->d->channel_map[c] == EBUR128_DUAL_MONO) {
+ channel_sum *= 2.0;
+ }
+ sum += channel_sum;
+ }
+ sum /= (double) frames_per_block;
+ if (optional_output) {
+ *optional_output = sum;
+ return EBUR128_SUCCESS;
+ } else if (sum >= histogram_energy_boundaries[0]) {
+ if (st->d->use_histogram) {
+ ++st->d->block_energy_histogram[find_histogram_index(sum)];
+ } else {
+ struct ebur128_dq_entry* block;
+ block = (struct ebur128_dq_entry*) malloc(sizeof(struct ebur128_dq_entry));
+ if (!block) return EBUR128_ERROR_NOMEM;
+ block->z = sum;
+ SLIST_INSERT_HEAD(&st->d->block_list, block, entries);
+ }
+ return EBUR128_SUCCESS;
+ } else {
+ return EBUR128_SUCCESS;
+ }
+}
+
+int ebur128_set_channel(ebur128_state* st,
+ unsigned int channel_number,
+ int value) {
+ if (channel_number >= st->channels) {
+ return 1;
+ }
+ if (value == EBUR128_DUAL_MONO &&
+ (st->channels != 1 || channel_number != 0)) {
+ fprintf(stderr, "EBUR128_DUAL_MONO only works with mono files!\n");
+ return 1;
+ }
+ st->d->channel_map[channel_number] = value;
+ return 0;
+}
+
+int ebur128_change_parameters(ebur128_state* st,
+ unsigned int channels,
+ unsigned long samplerate) {
+ int errcode;
+ if (channels == st->channels &&
+ samplerate == st->samplerate) {
+ return 2;
+ }
+ free(st->d->audio_data);
+ st->d->audio_data = NULL;
+
+ if (channels != st->channels) {
+ unsigned int i;
+
+ free(st->d->channel_map); st->d->channel_map = NULL;
+ free(st->d->sample_peak); st->d->sample_peak = NULL;
+ free(st->d->true_peak); st->d->true_peak = NULL;
+ st->channels = channels;
+
+#ifdef USE_SPEEX_RESAMPLER
+ ebur128_destroy_resampler(st);
+ ebur128_init_resampler(st);
+#endif
+
+ errcode = ebur128_init_channel_map(st);
+ CHECK_ERROR(errcode, EBUR128_ERROR_NOMEM, exit)
+
+ st->d->sample_peak = (double*) malloc(channels * sizeof(double));
+ CHECK_ERROR(!st->d->sample_peak, EBUR128_ERROR_NOMEM, exit)
+ st->d->true_peak = (double*) malloc(channels * sizeof(double));
+ CHECK_ERROR(!st->d->true_peak, EBUR128_ERROR_NOMEM, exit)
+ for (i = 0; i < channels; ++i) {
+ st->d->sample_peak[i] = 0.0;
+ st->d->true_peak[i] = 0.0;
+ }
+ }
+ if (samplerate != st->samplerate) {
+ st->samplerate = samplerate;
+ ebur128_init_filter(st);
+ }
+ if ((st->mode & EBUR128_MODE_S) == EBUR128_MODE_S) {
+ st->d->audio_data_frames = st->d->samples_in_100ms * 30;
+ } else if ((st->mode & EBUR128_MODE_M) == EBUR128_MODE_M) {
+ st->d->audio_data_frames = st->d->samples_in_100ms * 4;
+ } else {
+ return 1;
+ }
+ st->d->audio_data = (double*) malloc(st->d->audio_data_frames *
+ st->channels *
+ sizeof(double));
+ CHECK_ERROR(!st->d->audio_data, EBUR128_ERROR_NOMEM, exit)
+
+ /* the first block needs 400ms of audio data */
+ st->d->needed_frames = st->d->samples_in_100ms * 4;
+ /* start at the beginning of the buffer */
+ st->d->audio_data_index = 0;
+ /* reset short term frame counter */
+ st->d->short_term_frame_counter = 0;
+
+ return 0;
+
+exit:
+ return 1;
+}
+
+
+static int ebur128_energy_shortterm(ebur128_state* st, double* out);
+#define EBUR128_ADD_FRAMES(type) \
+int ebur128_add_frames_##type(ebur128_state* st, \
+ const type* src, size_t frames) { \
+ size_t src_index = 0; \
+ while (frames > 0) { \
+ if (frames >= st->d->needed_frames) { \
+ ebur128_filter_##type(st, src + src_index, st->d->needed_frames); \
+ src_index += st->d->needed_frames * st->channels; \
+ frames -= st->d->needed_frames; \
+ st->d->audio_data_index += st->d->needed_frames * st->channels; \
+ /* calculate the new gating block */ \
+ if ((st->mode & EBUR128_MODE_I) == EBUR128_MODE_I) { \
+ if (ebur128_calc_gating_block(st, st->d->samples_in_100ms * 4, NULL)) {\
+ return EBUR128_ERROR_NOMEM; \
+ } \
+ } \
+ if ((st->mode & EBUR128_MODE_LRA) == EBUR128_MODE_LRA) { \
+ st->d->short_term_frame_counter += st->d->needed_frames; \
+ if (st->d->short_term_frame_counter == st->d->samples_in_100ms * 30) { \
+ struct ebur128_dq_entry* block; \
+ double st_energy; \
+ ebur128_energy_shortterm(st, &st_energy); \
+ if (st_energy >= histogram_energy_boundaries[0]) { \
+ if (st->d->use_histogram) { \
+ ++st->d->short_term_block_energy_histogram[ \
+ find_histogram_index(st_energy)];\
+ } else { \
+ block = (struct ebur128_dq_entry*) \
+ malloc(sizeof(struct ebur128_dq_entry)); \
+ if (!block) return EBUR128_ERROR_NOMEM; \
+ block->z = st_energy; \
+ SLIST_INSERT_HEAD(&st->d->short_term_block_list, block, entries);\
+ } \
+ } \
+ st->d->short_term_frame_counter = st->d->samples_in_100ms * 20; \
+ } \
+ } \
+ /* 100ms are needed for all blocks besides the first one */ \
+ st->d->needed_frames = st->d->samples_in_100ms; \
+ /* reset audio_data_index when buffer full */ \
+ if (st->d->audio_data_index == st->d->audio_data_frames * st->channels) {\
+ st->d->audio_data_index = 0; \
+ } \
+ } else { \
+ ebur128_filter_##type(st, src + src_index, frames); \
+ st->d->audio_data_index += frames * st->channels; \
+ if ((st->mode & EBUR128_MODE_LRA) == EBUR128_MODE_LRA) { \
+ st->d->short_term_frame_counter += frames; \
+ } \
+ st->d->needed_frames -= frames; \
+ frames = 0; \
+ } \
+ } \
+ return EBUR128_SUCCESS; \
+}
+EBUR128_ADD_FRAMES(short)
+EBUR128_ADD_FRAMES(int)
+EBUR128_ADD_FRAMES(float)
+EBUR128_ADD_FRAMES(double)
+
+static int ebur128_gated_loudness(ebur128_state** sts, size_t size,
+ double* out) {
+ struct ebur128_dq_entry* it;
+ double relative_threshold = 0.0;
+ double gated_loudness = 0.0;
+ size_t above_thresh_counter = 0;
+ size_t i, j, start_index;
+
+ for (i = 0; i < size; i++) {
+ if (sts[i] && (sts[i]->mode & EBUR128_MODE_I) != EBUR128_MODE_I) {
+ return EBUR128_ERROR_INVALID_MODE;
+ }
+ }
+
+ for (i = 0; i < size; i++) {
+ if (!sts[i]) continue;
+ if (sts[i]->d->use_histogram) {
+ for (j = 0; j < 1000; ++j) {
+ relative_threshold += sts[i]->d->block_energy_histogram[j] *
+ histogram_energies[j];
+ above_thresh_counter += sts[i]->d->block_energy_histogram[j];
+ }
+ } else {
+ SLIST_FOREACH(it, &sts[i]->d->block_list, entries) {
+ ++above_thresh_counter;
+ relative_threshold += it->z;
+ }
+ }
+ }
+ if (!above_thresh_counter) {
+ *out = -HUGE_VAL;
+ return EBUR128_SUCCESS;
+ }
+ relative_threshold /= (double) above_thresh_counter;
+ relative_threshold *= relative_gate_factor;
+ above_thresh_counter = 0;
+ if (relative_threshold < histogram_energy_boundaries[0]) {
+ start_index = 0;
+ } else {
+ start_index = find_histogram_index(relative_threshold);
+ if (relative_threshold > histogram_energies[start_index]) {
+ ++start_index;
+ }
+ }
+ for (i = 0; i < size; i++) {
+ if (!sts[i]) continue;
+ if (sts[i]->d->use_histogram) {
+ for (j = start_index; j < 1000; ++j) {
+ gated_loudness += sts[i]->d->block_energy_histogram[j] *
+ histogram_energies[j];
+ above_thresh_counter += sts[i]->d->block_energy_histogram[j];
+ }
+ } else {
+ SLIST_FOREACH(it, &sts[i]->d->block_list, entries) {
+ if (it->z >= relative_threshold) {
+ ++above_thresh_counter;
+ gated_loudness += it->z;
+ }
+ }
+ }
+ }
+ if (!above_thresh_counter) {
+ *out = -HUGE_VAL;
+ return EBUR128_SUCCESS;
+ }
+ gated_loudness /= (double) above_thresh_counter;
+ *out = ebur128_energy_to_loudness(gated_loudness);
+ return EBUR128_SUCCESS;
+}
+
+int ebur128_loudness_global(ebur128_state* st, double* out) {
+ return ebur128_gated_loudness(&st, 1, out);
+}
+
+int ebur128_loudness_global_multiple(ebur128_state** sts, size_t size,
+ double* out) {
+ return ebur128_gated_loudness(sts, size, out);
+}
+
+static int ebur128_energy_in_interval(ebur128_state* st,
+ size_t interval_frames,
+ double* out) {
+ if (interval_frames > st->d->audio_data_frames) {
+ return EBUR128_ERROR_INVALID_MODE;
+ }
+ ebur128_calc_gating_block(st, interval_frames, out);
+ return EBUR128_SUCCESS;
+}
+
+static int ebur128_energy_shortterm(ebur128_state* st, double* out) {
+ return ebur128_energy_in_interval(st, st->d->samples_in_100ms * 30, out);
+}
+
+int ebur128_loudness_momentary(ebur128_state* st, double* out) {
+ double energy;
+ int error = ebur128_energy_in_interval(st, st->d->samples_in_100ms * 4,
+ &energy);
+ if (error) {
+ return error;
+ } else if (energy <= 0.0) {
+ *out = -HUGE_VAL;
+ return EBUR128_SUCCESS;
+ }
+ *out = ebur128_energy_to_loudness(energy);
+ return EBUR128_SUCCESS;
+}
+
+int ebur128_loudness_shortterm(ebur128_state* st, double* out) {
+ double energy;
+ int error = ebur128_energy_shortterm(st, &energy);
+ if (error) {
+ return error;
+ } else if (energy <= 0.0) {
+ *out = -HUGE_VAL;
+ return EBUR128_SUCCESS;
+ }
+ *out = ebur128_energy_to_loudness(energy);
+ return EBUR128_SUCCESS;
+}
+
+static int ebur128_double_cmp(const void *p1, const void *p2) {
+ const double* d1 = (const double*) p1;
+ const double* d2 = (const double*) p2;
+ return (*d1 > *d2) - (*d1 < *d2);
+}
+
+/* EBU - TECH 3342 */
+int ebur128_loudness_range_multiple(ebur128_state** sts, size_t size,
+ double* out) {
+ size_t i, j;
+ struct ebur128_dq_entry* it;
+ double* stl_vector;
+ size_t stl_size;
+ double* stl_relgated;
+ size_t stl_relgated_size;
+ double stl_power, stl_integrated;
+ /* High and low percentile energy */
+ double h_en, l_en;
+ int use_histogram = 0;
+
+ for (i = 0; i < size; ++i) {
+ if (sts[i]) {
+ if ((sts[i]->mode & EBUR128_MODE_LRA) != EBUR128_MODE_LRA) {
+ return EBUR128_ERROR_INVALID_MODE;
+ }
+ if (i == 0 && sts[i]->mode & EBUR128_MODE_HISTOGRAM) {
+ use_histogram = 1;
+ } else if (use_histogram != !!(sts[i]->mode & EBUR128_MODE_HISTOGRAM)) {
+ return EBUR128_ERROR_INVALID_MODE;
+ }
+ }
+ }
+
+ if (use_histogram) {
+ unsigned long hist[1000] = { 0 };
+ size_t percentile_low, percentile_high;
+ size_t index;
+
+ stl_size = 0;
+ stl_power = 0.0;
+ for (i = 0; i < size; ++i) {
+ if (!sts[i]) continue;
+ for (j = 0; j < 1000; ++j) {
+ hist[j] += sts[i]->d->short_term_block_energy_histogram[j];
+ stl_size += sts[i]->d->short_term_block_energy_histogram[j];
+ stl_power += sts[i]->d->short_term_block_energy_histogram[j]
+ * histogram_energies[j];
+ }
+ }
+ if (!stl_size) {
+ *out = 0.0;
+ return EBUR128_SUCCESS;
+ }
+
+ stl_power /= stl_size;
+ stl_integrated = minus_twenty_decibels * stl_power;
+
+ if (stl_integrated < histogram_energy_boundaries[0]) {
+ index = 0;
+ } else {
+ index = find_histogram_index(stl_integrated);
+ if (stl_integrated > histogram_energies[index]) {
+ ++index;
+ }
+ }
+ stl_size = 0;
+ for (j = index; j < 1000; ++j) {
+ stl_size += hist[j];
+ }
+ if (!stl_size) {
+ *out = 0.0;
+ return EBUR128_SUCCESS;
+ }
+
+ percentile_low = (size_t) ((stl_size - 1) * 0.1 + 0.5);
+ percentile_high = (size_t) ((stl_size - 1) * 0.95 + 0.5);
+
+ stl_size = 0;
+ j = index;
+ while (stl_size <= percentile_low) {
+ stl_size += hist[j++];
+ }
+ l_en = histogram_energies[j - 1];
+ while (stl_size <= percentile_high) {
+ stl_size += hist[j++];
+ }
+ h_en = histogram_energies[j - 1];
+ *out = ebur128_energy_to_loudness(h_en) - ebur128_energy_to_loudness(l_en);
+ return EBUR128_SUCCESS;
+
+ } else {
+ stl_size = 0;
+ for (i = 0; i < size; ++i) {
+ if (!sts[i]) continue;
+ SLIST_FOREACH(it, &sts[i]->d->short_term_block_list, entries) {
+ ++stl_size;
+ }
+ }
+ if (!stl_size) {
+ *out = 0.0;
+ return EBUR128_SUCCESS;
+ }
+ stl_vector = (double*) malloc(stl_size * sizeof(double));
+ if (!stl_vector)
+ return EBUR128_ERROR_NOMEM;
+
+ for (j = 0, i = 0; i < size; ++i) {
+ if (!sts[i]) continue;
+ SLIST_FOREACH(it, &sts[i]->d->short_term_block_list, entries) {
+ stl_vector[j] = it->z;
+ ++j;
+ }
+ }
+ qsort(stl_vector, stl_size, sizeof(double), ebur128_double_cmp);
+ stl_power = 0.0;
+ for (i = 0; i < stl_size; ++i) {
+ stl_power += stl_vector[i];
+ }
+ stl_power /= (double) stl_size;
+ stl_integrated = minus_twenty_decibels * stl_power;
+
+ stl_relgated = stl_vector;
+ stl_relgated_size = stl_size;
+ while (stl_relgated_size > 0 && *stl_relgated < stl_integrated) {
+ ++stl_relgated;
+ --stl_relgated_size;
+ }
+
+ if (stl_relgated_size) {
+ h_en = stl_relgated[(size_t) ((stl_relgated_size - 1) * 0.95 + 0.5)];
+ l_en = stl_relgated[(size_t) ((stl_relgated_size - 1) * 0.1 + 0.5)];
+ free(stl_vector);
+ *out = ebur128_energy_to_loudness(h_en) - ebur128_energy_to_loudness(l_en);
+ return EBUR128_SUCCESS;
+ } else {
+ free(stl_vector);
+ *out = 0.0;
+ return EBUR128_SUCCESS;
+ }
+ }
+}
+
+int ebur128_loudness_range(ebur128_state* st, double* out) {
+ return ebur128_loudness_range_multiple(&st, 1, out);
+}
+
+int ebur128_sample_peak(ebur128_state* st,
+ unsigned int channel_number,
+ double* out) {
+ if ((st->mode & EBUR128_MODE_SAMPLE_PEAK) != EBUR128_MODE_SAMPLE_PEAK) {
+ return EBUR128_ERROR_INVALID_MODE;
+ } else if (channel_number >= st->channels) {
+ return EBUR128_ERROR_INVALID_CHANNEL_INDEX;
+ }
+ *out = st->d->sample_peak[channel_number];
+ return EBUR128_SUCCESS;
+}
+
+#ifdef USE_SPEEX_RESAMPLER
+int ebur128_true_peak(ebur128_state* st,
+ unsigned int channel_number,
+ double* out) {
+ if ((st->mode & EBUR128_MODE_TRUE_PEAK) != EBUR128_MODE_TRUE_PEAK) {
+ return EBUR128_ERROR_INVALID_MODE;
+ } else if (channel_number >= st->channels) {
+ return EBUR128_ERROR_INVALID_CHANNEL_INDEX;
+ }
+ *out = st->d->true_peak[channel_number] > st->d->sample_peak[channel_number]
+ ? st->d->true_peak[channel_number]
+ : st->d->sample_peak[channel_number];
+ return EBUR128_SUCCESS;
+}
+#endif
--- /dev/null
+/* See COPYING file for copyright and license details. */
+
+#ifndef EBUR128_H_
+#define EBUR128_H_
+
+/** \file ebur128.h
+ * \brief libebur128 - a library for loudness measurement according to
+ * the EBU R128 standard.
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define EBUR128_VERSION_MAJOR 1
+#define EBUR128_VERSION_MINOR 0
+#define EBUR128_VERSION_PATCH 1
+
+#include <stddef.h> /* for size_t */
+
+/** \enum channel
+ * Use these values when setting the channel map with ebur128_set_channel().
+ */
+enum channel {
+ EBUR128_UNUSED = 0, /**< unused channel (for example LFE channel) */
+ EBUR128_LEFT, /**< left channel */
+ EBUR128_RIGHT, /**< right channel */
+ EBUR128_CENTER, /**< center channel */
+ EBUR128_LEFT_SURROUND, /**< left surround channel */
+ EBUR128_RIGHT_SURROUND, /**< right surround channel */
+ EBUR128_DUAL_MONO /**< a channel that is counted twice */
+};
+
+/** \enum error
+ * Error return values.
+ */
+enum error {
+ EBUR128_SUCCESS = 0,
+ EBUR128_ERROR_NOMEM,
+ EBUR128_ERROR_INVALID_MODE,
+ EBUR128_ERROR_INVALID_CHANNEL_INDEX,
+ EBUR128_ERROR_NO_CHANGE
+};
+
+/** \enum mode
+ * Use these values in ebur128_init (or'ed). Try to use the lowest possible
+ * modes that suit your needs, as performance will be better.
+ */
+enum mode {
+ /** can call ebur128_loudness_momentary */
+ EBUR128_MODE_M = (1 << 0),
+ /** can call ebur128_loudness_shortterm */
+ EBUR128_MODE_S = (1 << 1) | EBUR128_MODE_M,
+ /** can call ebur128_gated_loudness_* */
+ EBUR128_MODE_I = (1 << 2) | EBUR128_MODE_M,
+ /** can call ebur128_loudness_range */
+ EBUR128_MODE_LRA = (1 << 3) | EBUR128_MODE_S,
+ /** can call ebur128_sample_peak */
+ EBUR128_MODE_SAMPLE_PEAK = (1 << 4) | EBUR128_MODE_M,
+ /** can call ebur128_true_peak */
+ EBUR128_MODE_TRUE_PEAK = (1 << 5) | EBUR128_MODE_M
+ | EBUR128_MODE_SAMPLE_PEAK,
+ /** uses histogram algorithm to calculate loudness */
+ EBUR128_MODE_HISTOGRAM = (1 << 6)
+};
+
+/** forward declaration of ebur128_state_internal */
+struct ebur128_state_internal;
+
+/** \brief Contains information about the state of a loudness measurement.
+ *
+ * You should not need to modify this struct directly.
+ */
+typedef struct {
+ int mode; /**< The current mode. */
+ unsigned int channels; /**< The number of channels. */
+ unsigned long samplerate; /**< The sample rate. */
+ struct ebur128_state_internal* d; /**< Internal state. */
+} ebur128_state;
+
+/** \brief Get library version number. Do not pass null pointers here.
+ *
+ * @param major major version number of library
+ * @param minor minor version number of library
+ * @param patch patch version number of library
+ */
+void ebur128_get_version(int* major, int* minor, int* patch);
+
+/** \brief Initialize library state.
+ *
+ * @param channels the number of channels.
+ * @param samplerate the sample rate.
+ * @param mode see the mode enum for possible values.
+ * @return an initialized library state.
+ */
+ebur128_state* ebur128_init(unsigned int channels,
+ unsigned long samplerate,
+ int mode);
+
+/** \brief Destroy library state.
+ *
+ * @param st pointer to a library state.
+ */
+void ebur128_destroy(ebur128_state** st);
+
+/** \brief Set channel type.
+ *
+ * The default is:
+ * - 0 -> EBUR128_LEFT
+ * - 1 -> EBUR128_RIGHT
+ * - 2 -> EBUR128_CENTER
+ * - 3 -> EBUR128_UNUSED
+ * - 4 -> EBUR128_LEFT_SURROUND
+ * - 5 -> EBUR128_RIGHT_SURROUND
+ *
+ * @param st library state.
+ * @param channel_number zero based channel index.
+ * @param value channel type from the "channel" enum.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_CHANNEL_INDEX if invalid channel index.
+ */
+int ebur128_set_channel(ebur128_state* st,
+ unsigned int channel_number,
+ int value);
+
+/** \brief Change library parameters.
+ *
+ * Note that the channel map will be reset when setting a different number of
+ * channels. The current unfinished block will be lost.
+ *
+ * @param st library state.
+ * @param channels new number of channels.
+ * @param samplerate new sample rate.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_NOMEM on memory allocation error. The state will be
+ * invalid and must be destroyed.
+ * - EBUR128_ERROR_NO_CHANGE if channels and sample rate were not changed.
+ */
+int ebur128_change_parameters(ebur128_state* st,
+ unsigned int channels,
+ unsigned long samplerate);
+
+/** \brief Add frames to be processed.
+ *
+ * @param st library state.
+ * @param src array of source frames. Channels must be interleaved.
+ * @param frames number of frames. Not number of samples!
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_NOMEM on memory allocation error.
+ */
+int ebur128_add_frames_short(ebur128_state* st,
+ const short* src,
+ size_t frames);
+/** \brief See \ref ebur128_add_frames_short */
+int ebur128_add_frames_int(ebur128_state* st,
+ const int* src,
+ size_t frames);
+/** \brief See \ref ebur128_add_frames_short */
+int ebur128_add_frames_float(ebur128_state* st,
+ const float* src,
+ size_t frames);
+/** \brief See \ref ebur128_add_frames_short */
+int ebur128_add_frames_double(ebur128_state* st,
+ const double* src,
+ size_t frames);
+
+/** \brief Get global integrated loudness in LUFS.
+ *
+ * @param st library state.
+ * @param out integrated loudness in LUFS. -HUGE_VAL if result is negative
+ * infinity.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_I" has not been set.
+ */
+int ebur128_loudness_global(ebur128_state* st, double* out);
+/** \brief Get global integrated loudness in LUFS across multiple instances.
+ *
+ * @param sts array of library states.
+ * @param size length of sts
+ * @param out integrated loudness in LUFS. -HUGE_VAL if result is negative
+ * infinity.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_I" has not been set.
+ */
+int ebur128_loudness_global_multiple(ebur128_state** sts,
+ size_t size,
+ double* out);
+
+/** \brief Get momentary loudness (last 400ms) in LUFS.
+ *
+ * @param st library state.
+ * @param out momentary loudness in LUFS. -HUGE_VAL if result is negative
+ * infinity.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ */
+int ebur128_loudness_momentary(ebur128_state* st, double* out);
+/** \brief Get short-term loudness (last 3s) in LUFS.
+ *
+ * @param st library state.
+ * @param out short-term loudness in LUFS. -HUGE_VAL if result is negative
+ * infinity.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_S" has not been set.
+ */
+int ebur128_loudness_shortterm(ebur128_state* st, double* out);
+
+/** \brief Get loudness range (LRA) of programme in LU.
+ *
+ * Calculates loudness range according to EBU 3342.
+ *
+ * @param st library state.
+ * @param out loudness range (LRA) in LU. Will not be changed in case of
+ * error. EBUR128_ERROR_NOMEM or EBUR128_ERROR_INVALID_MODE will be
+ * returned in this case.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_NOMEM in case of memory allocation error.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_LRA" has not been set.
+ */
+int ebur128_loudness_range(ebur128_state* st, double* out);
+/** \brief Get loudness range (LRA) in LU across multiple instances.
+ *
+ * Calculates loudness range according to EBU 3342.
+ *
+ * @param sts array of library states.
+ * @param size length of sts
+ * @param out loudness range (LRA) in LU. Will not be changed in case of
+ * error. EBUR128_ERROR_NOMEM or EBUR128_ERROR_INVALID_MODE will be
+ * returned in this case.
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_NOMEM in case of memory allocation error.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_LRA" has not been set.
+ */
+int ebur128_loudness_range_multiple(ebur128_state** sts,
+ size_t size,
+ double* out);
+
+/** \brief Get maximum sample peak of selected channel in float format.
+ *
+ * @param st library state
+ * @param channel_number channel to analyse
+ * @param out maximum sample peak in float format (1.0 is 0 dBFS)
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_SAMPLE_PEAK" has not
+ * been set.
+ * - EBUR128_ERROR_INVALID_CHANNEL_INDEX if invalid channel index.
+ */
+int ebur128_sample_peak(ebur128_state* st,
+ unsigned int channel_number,
+ double* out);
+
+/** \brief Get maximum true peak of selected channel in float format.
+ *
+ * Uses an implementation defined algorithm to calculate the true peak. Do not
+ * try to compare resulting values across different versions of the library,
+ * as the algorithm may change.
+ *
+ * The current implementation uses the Speex resampler with quality level 8 to
+ * calculate true peak. Will oversample 4x for sample rates < 96000 Hz, 2x for
+ * sample rates < 192000 Hz and leave the signal unchanged for 192000 Hz.
+ *
+ * @param st library state
+ * @param channel_number channel to analyse
+ * @param out maximum true peak in float format (1.0 is 0 dBFS)
+ * @return
+ * - EBUR128_SUCCESS on success.
+ * - EBUR128_ERROR_INVALID_MODE if mode "EBUR128_MODE_TRUE_PEAK" has not
+ * been set.
+ * - EBUR128_ERROR_INVALID_CHANNEL_INDEX if invalid channel index.
+ */
+int ebur128_true_peak(ebur128_state* st,
+ unsigned int channel_number,
+ double* out);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* EBUR128_H_ */
extern mlt_filter filter_affine_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
extern mlt_filter filter_charcoal_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
extern mlt_filter filter_dynamictext_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
+extern mlt_filter filter_loudness_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
extern mlt_filter filter_invert_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
extern mlt_filter filter_sepia_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
extern mlt_producer producer_blipflash_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
MLT_REGISTER( filter_type, "charcoal", filter_charcoal_init );
MLT_REGISTER( filter_type, "dynamictext", filter_dynamictext_init );
MLT_REGISTER( filter_type, "invert", filter_invert_init );
+ MLT_REGISTER( filter_type, "loudness", filter_loudness_init );
MLT_REGISTER( filter_type, "sepia", filter_sepia_init );
MLT_REGISTER( producer_type, "blipflash", producer_blipflash_init );
MLT_REGISTER( producer_type, "count", producer_count_init );
MLT_REGISTER_METADATA( filter_type, "charcoal", metadata, "filter_charcoal.yml" );
MLT_REGISTER_METADATA( filter_type, "dynamictext", metadata, "filter_dynamictext.yml" );
MLT_REGISTER_METADATA( filter_type, "invert", metadata, "filter_invert.yml" );
+ MLT_REGISTER_METADATA( filter_type, "loudness", metadata, "filter_loudness.yml" );
MLT_REGISTER_METADATA( filter_type, "sepia", metadata, "filter_sepia.yml" );
MLT_REGISTER_METADATA( producer_type, "blipflash", metadata, "producer_blipflash.yml" );
MLT_REGISTER_METADATA( producer_type, "count", metadata, "producer_count.yml" );
--- /dev/null
+/*
+ * filter_ebur128.c -- normalize audio according to EBU R128
+ * Copyright (C) 2014 Brian Matherly <pez4brian@yahoo.com>
+ * Author: Brian Matherly <pez4brian@yahoo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <framework/mlt.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "ebur128/ebur128.h"
+
+#define MAX_RESULT_SIZE 512
+
+typedef struct
+{
+ ebur128_state* state;
+} analyze_data;
+
+typedef struct
+{
+ double in_loudness;
+ double in_range;
+ double in_peak;
+ double coeff;
+} apply_data;
+
+typedef struct
+{
+ analyze_data* analyze;
+ apply_data* apply;
+ mlt_position last_position;
+} private_data;
+
+static void destroy_analyze_data( mlt_filter filter )
+{
+ private_data* private = (private_data*)filter->child;
+ ebur128_destroy( &private->analyze->state );
+ free( private->analyze );
+ private->analyze = NULL;
+}
+
+static void init_analyze_data( mlt_filter filter, int channels, int samplerate )
+{
+ private_data* private = (private_data*)filter->child;
+ private->analyze = (analyze_data*)calloc( 1, sizeof(analyze_data) );
+ private->analyze->state = ebur128_init( (unsigned int)channels, (unsigned long)samplerate, EBUR128_MODE_I | EBUR128_MODE_LRA | EBUR128_MODE_SAMPLE_PEAK );
+ private->last_position = 0;
+}
+
+static void destroy_apply_data( mlt_filter filter )
+{
+ private_data* private = (private_data*)filter->child;
+ free( private->apply );
+ private->apply = NULL;
+}
+
+static void init_apply_data( mlt_filter filter )
+{
+ private_data* private = (private_data*)filter->child;
+ mlt_properties properties = MLT_FILTER_PROPERTIES( filter );
+ char* results = mlt_properties_get( properties, "results" );
+ int scan_return = 0;
+
+ private->apply = (apply_data*)calloc( 1, sizeof(apply_data) );
+
+ scan_return = sscanf( results,"L: %lf\tR: %lf\tP %lf\n", &private->apply->in_loudness, &private->apply->in_range, &private->apply->in_peak );
+ if( scan_return != 3 )
+ {
+ mlt_log_error( MLT_FILTER_SERVICE( filter ), "Unable to load results: %s\n", results );
+ destroy_apply_data( filter );
+ return;
+ }
+ else
+ {
+ double target_db = mlt_properties_get_double( properties, "program" );
+ double delta_db = target_db - private->apply->in_loudness;
+ private->apply->coeff = delta_db > -90.0f ? powf(10.0f, delta_db * 0.05f) : 0.0f;
+ mlt_log_info( MLT_FILTER_SERVICE( filter ), "Loaded Results: L: %lf\tR: %lf\tP %lf\n", private->apply->in_loudness, private->apply->in_range, private->apply->in_peak );
+ mlt_log_info( MLT_FILTER_SERVICE( filter ), "Coefficient: %lf\n", private->apply->coeff );
+ }
+}
+
+static void analyze( mlt_filter filter, mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+{
+ private_data* private = (private_data*)filter->child;
+ mlt_position pos = mlt_filter_get_position( filter, frame );
+
+ // If any frames are skipped, analysis data will be incomplete.
+ if( private->analyze && pos != private->last_position + 1 )
+ {
+ mlt_log_error( MLT_FILTER_SERVICE(filter), "Analysis Failed: Bad frame sequence\n" );
+ destroy_analyze_data( filter );
+ }
+
+ // Analyze Audio
+ if( !private->analyze && pos == 0 )
+ {
+ init_analyze_data( filter, *channels, *frequency );
+ }
+
+ if( private->analyze )
+ {
+ ebur128_add_frames_float( private->analyze->state, *buffer, *samples );
+
+ if ( pos + 1 == mlt_filter_get_length2( filter, frame ) )
+ {
+ mlt_properties properties = MLT_FILTER_PROPERTIES( filter );
+ double loudness = 0.0;
+ double range = 0.0;
+ double tmpPeak = 0.0;
+ double peak = 0.0;
+ int i = 0;
+ char result[MAX_RESULT_SIZE];
+ ebur128_loudness_global( private->analyze->state, &loudness );
+ ebur128_loudness_range( private->analyze->state, &range );
+
+ for ( i = 0; i < *channels; i++ )
+ {
+ ebur128_sample_peak( private->analyze->state, i, &tmpPeak );
+ if( tmpPeak > peak )
+ {
+ peak = tmpPeak;
+ }
+ }
+
+ snprintf( result, MAX_RESULT_SIZE, "L: %lf\tR: %lf\tP %lf\n", loudness, range, peak );
+ result[ MAX_RESULT_SIZE - 1 ] = '\0';
+ mlt_log_info( MLT_FILTER_SERVICE( filter ), "Stored results: %s", result );
+ mlt_properties_set( properties, "results", result );
+ destroy_analyze_data( filter );
+ }
+
+ private->last_position = pos;
+ }
+}
+
+static void apply( mlt_filter filter, mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+{
+ private_data* private = (private_data*)filter->child;
+
+ // Analyze Audio
+ if( !private->apply )
+ {
+ init_apply_data( filter );
+ }
+
+ if( private->apply )
+ {
+ float* p = *buffer;
+ int count = *samples * *channels;
+ while( count-- )
+ {
+ *p = *p * private->apply->coeff;
+ p++;
+ }
+ }
+}
+
+/** Get the audio.
+*/
+
+static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+{
+ mlt_filter filter = mlt_frame_pop_audio( frame );
+ mlt_properties properties = MLT_FILTER_PROPERTIES( filter );
+
+ mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
+
+ // Get the producer's audio
+ *format = mlt_audio_f32le;
+ mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+
+ char* results = mlt_properties_get( properties, "results" );
+ if( results && strcmp( results, "" ) )
+ {
+ apply( filter, frame, buffer, format, frequency, channels, samples );
+ }
+ else
+ {
+ analyze( filter, frame, buffer, format, frequency, channels, samples );
+ }
+
+ mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
+
+ return 0;
+}
+
+/** Filter processing.
+*/
+
+static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
+{
+ mlt_frame_push_audio( frame, filter );
+ mlt_frame_push_audio( frame, filter_get_audio );
+ return frame;
+}
+
+/** Destructor for the filter.
+*/
+
+static void filter_close( mlt_filter filter )
+{
+ private_data* private = (private_data*)filter->child;
+
+ if ( private )
+ {
+ if ( private->analyze )
+ {
+ destroy_analyze_data( filter );
+ }
+ free( private );
+ }
+ filter->child = NULL;
+ filter->close = NULL;
+ filter->parent.close = NULL;
+ mlt_service_close( &filter->parent );
+}
+
+/** Constructor for the filter.
+*/
+
+mlt_filter filter_loudness_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
+{
+ mlt_filter filter = mlt_filter_new( );
+ private_data* data = (private_data*)calloc( 1, sizeof(private_data) );
+
+ if ( filter && data )
+ {
+ mlt_properties properties = MLT_FILTER_PROPERTIES( filter );
+ mlt_properties_set( properties, "program", "-23.0" );
+
+ data->analyze = NULL;
+
+ filter->close = filter_close;
+ filter->process = filter_process;
+ filter->child = data;
+ }
+ else
+ {
+ if( filter )
+ {
+ mlt_filter_close( filter );
+ filter = NULL;
+ }
+
+ if( data )
+ {
+ free( data );
+ }
+ }
+
+ return filter;
+}
--- /dev/null
+schema_version: 0.1
+type: filter
+identifier: loudness
+title: Loudness
+version: 1
+copyright: Brian Matherly <pez4brian@yahoo.com>
+creator: Brian Matherly <pez4brian@yahoo.com>
+license: LGPLv2.1
+language: en
+tags:
+ - Audio
+description: Correct audio loudness as recommended by EBU R128.
+notes: >
+ This filter requires two passes. The first pass performs analysis and stores
+ the result in the "results" property. The second pass applies the results to
+ the audio in order to achieve the desired loudness over the range of the
+ filter.
+
+parameters:
+ - identifier: results
+ title: Analysis Results
+ type: string
+ description: >
+ Set after analysis. Used during application.
+ Loudness information about the original audio.
+ When results are not supplied, the filter computes the results and stores
+ them in this property when the last frame has been processed.
+ mutable: no
+
+ - identifier: program
+ title: Target Program Loudness
+ type: float
+ description: >
+ Used during application.
+ The target program loudness in LUFS (Loudness Units Full Scale).
+ readonly: no
+ mutable: no
+ default: -23.0
+ minimum: -50.0
+ maximum: -10.0
+ unit: LUFS
projects / mlt / commitdiff