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avfilter/f_ebur128: add all sample rates support

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The magic constants come from the unofficial "ITU-R BS.1770-1 filter
specifications"¹ by Raiden (libebur128) which relies on "Parameter
Quantization in Direct-Form Recursive Audio Filters"² by Brian
Neunaber.

The constants seem to include a quantization bias, for example:
- Vb is supposed to be exactly √Vh in a high shelf filter
- the Pre-filter Gain should likely be 4dB
- Pre Q and RLB Q are respectively very close to √½ and ½

Those are not adjusted to prevent the values from drifting away from
the official specifications.

An alternative to this approach would be to requantize on the fly as
proposed by pbelkner³, where the 48kHz code path would use the exact
specifications constants while derivating constants for other
frequencies.

[1]: https://www.scribd.com/document/49991813/ITU-R-BS-1770-1-filters
[2]: https://www.scribd.com/document/6531763/Direct-Form-Filter-Parameter-Quantization
[3]: https://hydrogenaud.io/index.php?topic=86116.msg740092#msg740092
This commit is contained in:
Paul B Mahol 2021-03-04 14:01:39 +01:00
parent a2a7547b2f
commit 274112c88d
1 changed files with 49 additions and 27 deletions
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76
libavfilter/f_ebur128.c
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@ -24,7 +24,6 @@
* @see http://tech.ebu.ch/loudness
* @see https://www.youtube.com/watch?v=iuEtQqC-Sqo "EBU R128 Introduction - Florian Camerer"
* @todo implement start/stop/reset through filter command injection
* @todo support other frequencies to avoid resampling
*/
#include <math.h>
@ -45,20 +44,6 @@
#define MAX_CHANNELS 63
/* pre-filter coefficients */
#define PRE_B0 1.53512485958697
#define PRE_B1 -2.69169618940638
#define PRE_B2 1.19839281085285
#define PRE_A1 -1.69065929318241
#define PRE_A2 0.73248077421585
/* RLB-filter coefficients */
#define RLB_B0 1.0
#define RLB_B1 -2.0
#define RLB_B2 1.0
#define RLB_A1 -1.99004745483398
#define RLB_A2 0.99007225036621
#define ABS_THRES -70 ///< silence gate: we discard anything below this absolute (LUFS) threshold
#define ABS_UP_THRES 10 ///< upper loud limit to consider (ABS_THRES being the minimum)
#define HIST_GRAIN 100 ///< defines histogram precision
@ -80,6 +65,7 @@ struct hist_entry {
struct integrator {
double *cache[MAX_CHANNELS]; ///< window of filtered samples (N ms)
int cache_pos; ///< focus on the last added bin in the cache array
int cache_size;
double sum[MAX_CHANNELS]; ///< sum of the last N ms filtered samples (cache content)
int filled; ///< 1 if the cache is completely filled, 0 otherwise
double rel_threshold; ///< relative threshold
@ -128,9 +114,11 @@ typedef struct EBUR128Context {
double x[MAX_CHANNELS * 3]; ///< 3 input samples cache for each channel
double y[MAX_CHANNELS * 3]; ///< 3 pre-filter samples cache for each channel
double z[MAX_CHANNELS * 3]; ///< 3 RLB-filter samples cache for each channel
double pre_b[3]; ///< pre-filter numerator coefficients
double pre_a[3]; ///< pre-filter denominator coefficients
double rlb_b[3]; ///< rlb-filter numerator coefficients
double rlb_a[3]; ///< rlb-filter denominator coefficients
#define I400_BINS (48000 * 4 / 10)
#define I3000_BINS (48000 * 3)
struct integrator i400; ///< 400ms integrator, used for Momentary loudness (M), and Integrated loudness (I)
struct integrator i3000; ///< 3s integrator, used for Short term loudness (S), and Loudness Range (LRA)
@ -388,6 +376,35 @@ static int config_audio_input(AVFilterLink *inlink)
AVFilterContext *ctx = inlink->dst;
EBUR128Context *ebur128 = ctx->priv;
/* Unofficial reversed parametrization of PRE
* and RLB from 48kHz */
double f0 = 1681.974450955533;
double G = 3.999843853973347;
double Q = 0.7071752369554196;
double K = tan(M_PI * f0 / (double)inlink->sample_rate);
double Vh = pow(10.0, G / 20.0);
double Vb = pow(Vh, 0.4996667741545416);
double a0 = 1.0 + K / Q + K * K;
ebur128->pre_b[0] = (Vh + Vb * K / Q + K * K) / a0;
ebur128->pre_b[1] = 2.0 * (K * K - Vh) / a0;
ebur128->pre_b[2] = (Vh - Vb * K / Q + K * K) / a0;
ebur128->pre_a[1] = 2.0 * (K * K - 1.0) / a0;
ebur128->pre_a[2] = (1.0 - K / Q + K * K) / a0;
f0 = 38.13547087602444;
Q = 0.5003270373238773;
K = tan(M_PI * f0 / (double)inlink->sample_rate);
ebur128->rlb_b[0] = 1.0;
ebur128->rlb_b[1] = -2.0;
ebur128->rlb_b[2] = 1.0;
ebur128->rlb_a[1] = 2.0 * (K * K - 1.0) / (1.0 + K / Q + K * K);
ebur128->rlb_a[2] = (1.0 - K / Q + K * K) / (1.0 + K / Q + K * K);
/* Force 100ms framing in case of metadata injection: the frames must have
* a granularity of the window overlap to be accurately exploited.
* As for the true peaks mode, it just simplifies the resampling buffer
@ -418,6 +435,9 @@ static int config_audio_output(AVFilterLink *outlink)
if (!ebur128->ch_weighting)
return AVERROR(ENOMEM);
#define I400_BINS (outlink->sample_rate * 4 / 10)
#define I3000_BINS (outlink->sample_rate * 3)
for (i = 0; i < nb_channels; i++) {
/* channel weighting */
const uint64_t chl = av_channel_layout_extract_channel(outlink->channel_layout, i);
@ -433,8 +453,10 @@ static int config_audio_output(AVFilterLink *outlink)
continue;
/* bins buffer for the two integration window (400ms and 3s) */
ebur128->i400.cache[i] = av_calloc(I400_BINS, sizeof(*ebur128->i400.cache[0]));
ebur128->i3000.cache[i] = av_calloc(I3000_BINS, sizeof(*ebur128->i3000.cache[0]));
ebur128->i400.cache_size = I400_BINS;
ebur128->i3000.cache_size = I3000_BINS;
ebur128->i400.cache[i] = av_calloc(ebur128->i400.cache_size, sizeof(*ebur128->i400.cache[0]));
ebur128->i3000.cache[i] = av_calloc(ebur128->i3000.cache_size, sizeof(*ebur128->i3000.cache[0]));
if (!ebur128->i400.cache[i] || !ebur128->i3000.cache[i])
return AVERROR(ENOMEM);
}
@ -613,7 +635,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
#define MOVE_TO_NEXT_CACHED_ENTRY(time) do { \
ebur128->i##time.cache_pos++; \
if (ebur128->i##time.cache_pos == I##time##_BINS) { \
if (ebur128->i##time.cache_pos == \
ebur128->i##time.cache_size) { \
ebur128->i##time.filled = 1; \
ebur128->i##time.cache_pos = 0; \
} \
@ -634,20 +657,20 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
continue;
/* Y[i] = X[i]*b0 + X[i-1]*b1 + X[i-2]*b2 - Y[i-1]*a1 - Y[i-2]*a2 */
#define FILTER(Y, X, name) do { \
#define FILTER(Y, X, NUM, DEN) do { \
double *dst = ebur128->Y + ch*3; \
double *src = ebur128->X + ch*3; \
dst[2] = dst[1]; \
dst[1] = dst[0]; \
dst[0] = src[0]*name##_B0 + src[1]*name##_B1 + src[2]*name##_B2 \
- dst[1]*name##_A1 - dst[2]*name##_A2; \
dst[0] = src[0]*NUM[0] + src[1]*NUM[1] + src[2]*NUM[2] \
- dst[1]*DEN[1] - dst[2]*DEN[2]; \
} while (0)
// TODO: merge both filters in one?
FILTER(y, x, PRE); // apply pre-filter
FILTER(y, x, ebur128->pre_b, ebur128->pre_a); // apply pre-filter
ebur128->x[ch * 3 + 2] = ebur128->x[ch * 3 + 1];
ebur128->x[ch * 3 + 1] = ebur128->x[ch * 3 ];
FILTER(z, y, RLB); // apply RLB-filter
FILTER(z, y, ebur128->rlb_b, ebur128->rlb_a); // apply RLB-filter
bin = ebur128->z[ch * 3] * ebur128->z[ch * 3];
@ -896,7 +919,6 @@ static int query_formats(AVFilterContext *ctx)
int ret;
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_NONE };
static const int input_srate[] = {48000, -1}; // ITU-R BS.1770 provides coeff only for 48kHz
static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGB24, AV_PIX_FMT_NONE };
/* set optional output video format */
@ -920,7 +942,7 @@ static int query_formats(AVFilterContext *ctx)
(ret = ff_channel_layouts_ref(layouts, &outlink->incfg.channel_layouts)) < 0)
return ret;
formats = ff_make_format_list(input_srate);
formats = ff_all_samplerates();
if ((ret = ff_formats_ref(formats, &inlink->outcfg.samplerates)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->incfg.samplerates)) < 0)
return ret;