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avfilter/af_dynaudnorm: add slice threading support

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This commit is contained in:
Paul B Mahol 2022-11-04 17:59:28 +01:00
parent 4a672f1c0e
commit 369b7f2654
1 changed files with 64 additions and 22 deletions
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86
libavfilter/af_dynaudnorm.c
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@ -93,6 +93,11 @@ typedef struct DynamicAudioNormalizerContext {
AVFrame *window;
} DynamicAudioNormalizerContext;
typedef struct ThreadData {
AVFrame *in, *out;
int enabled;
} ThreadData;
#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
@ -521,6 +526,20 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
}
}
static int update_gain_histories(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
DynamicAudioNormalizerContext *s = ctx->priv;
AVFrame *analyze_frame = arg;
const int channels = s->channels;
const int start = (channels * jobnr) / nb_jobs;
const int end = (channels * (jobnr+1)) / nb_jobs;
for (int c = start; c < end; c++)
update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
return 0;
}
static inline double update_value(double new, double old, double aggressiveness)
{
av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
@ -655,8 +674,9 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
}
}
static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink, AVFrame **frame)
static int analyze_frame(AVFilterContext *ctx, AVFilterLink *outlink, AVFrame **frame)
{
DynamicAudioNormalizerContext *s = ctx->priv;
AVFrame *analyze_frame;
if (s->dc_correction || s->compress_factor > DBL_EPSILON) {
@ -716,34 +736,49 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
for (int c = 0; c < s->channels; c++)
update_gain_history(s, c, gain);
} else {
for (int c = 0; c < s->channels; c++)
update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
ff_filter_execute(ctx, update_gain_histories, analyze_frame, NULL,
FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
}
return 0;
}
static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in,
AVFrame *frame, int enabled)
static void amplify_channel(DynamicAudioNormalizerContext *s, AVFrame *in,
AVFrame *frame, int enabled, int 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];
double current_amplification_factor;
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];
double current_amplification_factor;
cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
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);
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);
dst_ptr[i] = src_ptr[i] * amplification_factor;
}
s->prev_amplification_factor[c] = current_amplification_factor;
dst_ptr[i] = src_ptr[i] * amplification_factor;
}
s->prev_amplification_factor[c] = current_amplification_factor;
}
static int amplify_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
DynamicAudioNormalizerContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int enabled = td->enabled;
const int channels = s->channels;
const int start = (channels * jobnr) / nb_jobs;
const int end = (channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++)
amplify_channel(s, in, out, enabled, ch);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
@ -751,6 +786,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterContext *ctx = inlink->dst;
DynamicAudioNormalizerContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
ThreadData td;
int ret;
while (((s->queue.available >= s->filter_size) ||
@ -773,7 +809,12 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_copy_props(out, in);
}
amplify_frame(s, in, out, is_enabled > 0.);
td.in = in;
td.out = out;
td.enabled = is_enabled > 0.;
ff_filter_execute(ctx, amplify_channels, &td, NULL,
FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
s->pts = out->pts + av_rescale_q(out->nb_samples, av_make_q(1, outlink->sample_rate),
outlink->time_base);
if (out != in)
@ -783,7 +824,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
return ret;
}
ret = analyze_frame(s, outlink, &in);
ret = analyze_frame(ctx, outlink, &in);
if (ret < 0)
return ret;
if (!s->eof) {
@ -940,6 +981,7 @@ const AVFilter ff_af_dynaudnorm = {
FILTER_OUTPUTS(avfilter_af_dynaudnorm_outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
.priv_class = &dynaudnorm_class,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = process_command,
};