Re: [FFmpeg-devel] One pass volume normalization (ebur128)

13 Jul 2013

On 7/13/13, Jan Ehrhardt <phpdev@ehrhardt.nl> wrote:
...
I am once again proposing a patch for one pass volume normalization base
on ebur128, as I see this still did not make it into FFMpeg 2.0. My
patch is heaveily based on Clement Boesch's proposal in
http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/159978
We have been using this patch now for more than 4 months and 1800+
videos of approximately 1 hour have been transcoded with it.
Part of our FFMpeg commandline reads as
-filter_complex \
"[0:v]setpts=PTS-STARTPTS[v0];[0:a]asetpts=PTS-STARTPTS,ebur128=metadata=1,volume=metadata=lavfi.r128.I,ebur128[a0]"
\
   -map [v0] -map [a0]
It uses the already present ebur128 meta injection to adjust the
volume on the fly. What would be the objection to move this into the
FFMpeg core, so I do not have to patch my FFMpeg every time I compile
a new one? I applied the patch below to FFMpeg Release/v.2.0.
Jan

diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c
index a2ac1e2..6372bb2 100644
--- a/libavfilter/af_volume.c
+++ b/libavfilter/af_volume.c
@@ -51,18 +51,24 @@ static const AVOption volume_options[] = {
         { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, {
.i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },
         { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, {
.i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },
         { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, {
.i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
+    { "metadata", "set the metadata key for loudness normalization",
OFFSET(metadata), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = A|F },
     { NULL },
 };
AVFILTER_DEFINE_CLASS(volume);
+static void set_fixed_volume(VolumeContext *vol, double volume)
+{
+    vol->volume_i = (int)(volume * 256 + 0.5);
+    vol->volume   = vol->volume_i / 256.0;
+}
+
 static av_cold int init(AVFilterContext *ctx)
 {
     VolumeContext *vol = ctx->priv;
if (vol->precision == PRECISION_FIXED) {
-        vol->volume_i = (int)(vol->volume * 256 + 0.5);
-        vol->volume   = vol->volume_i / 256.0;
+        set_fixed_volume(vol, vol->volume);
         av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB)
precision:fixed\n",
                vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
     } else {
@@ -171,13 +177,13 @@ static av_cold void volume_init(VolumeContext *vol)
switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
     case AV_SAMPLE_FMT_U8:
-        if (vol->volume_i < 0x1000000)
+        if (vol->volume_i < 0x1000000 && !vol->metadata)
             vol->scale_samples = scale_samples_u8_small;
         else
             vol->scale_samples = scale_samples_u8;
         break;
     case AV_SAMPLE_FMT_S16:
-        if (vol->volume_i < 0x10000)
+        if (vol->volume_i < 0x10000 && !vol->metadata)
             vol->scale_samples = scale_samples_s16_small;
         else
             vol->scale_samples = scale_samples_s16;
@@ -216,11 +222,30 @@ static int config_output(AVFilterLink *outlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
 {
-    VolumeContext *vol    = inlink->dst->priv;
-    AVFilterLink *outlink = inlink->dst->outputs[0];
+    AVFilterContext *ctx  = inlink->dst;
+    VolumeContext *vol    = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
     int nb_samples        = buf->nb_samples;
     AVFrame *out_buf;
+    if (vol->metadata) {
+        double loudness, new_volume, timestamp, mx;
+        AVDictionaryEntry *e;
+        mx = 20;
+        timestamp = (float)(1.0 * buf->pts / outlink->sample_rate);
+        mx = fmin(mx, timestamp);
+        e = av_dict_get(buf->metadata, vol->metadata, NULL, 0);
+        if (e) {
+            loudness = av_strtod(e->value, NULL);
+            if (loudness > -69) {
+                new_volume = fmax(-mx,fmin(mx,(-23 - loudness)));
+                av_log(NULL, AV_LOG_VERBOSE, "loudness=%f => %f =>
volume=%f\n",
+                    loudness, new_volume, pow(10, new_volume / 20));
+                set_fixed_volume(vol, pow(10, new_volume / 20));
+            }
+        }
+    }
+
     if (vol->volume == 1.0 || vol->volume_i == 256)
         return ff_filter_frame(outlink, buf);
@@ -269,6 +294,12 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
*buf)
     return ff_filter_frame(outlink, out_buf);
 }
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    VolumeContext *vol = ctx->priv;
+    av_opt_free(vol);
+}
+
 static const AVFilterPad avfilter_af_volume_inputs[] = {
     {
         .name           = "default",
@@ -294,6 +325,7 @@ AVFilter avfilter_af_volume = {
     .priv_size      = sizeof(VolumeContext),
     .priv_class     = &volume_class,
     .init           = init,
+    .uninit         = uninit,
     .inputs         = avfilter_af_volume_inputs,
     .outputs        = avfilter_af_volume_outputs,
     .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
diff --git a/libavfilter/af_volume.h b/libavfilter/af_volume.h
index bd7932e..4deca9c 100644
--- a/libavfilter/af_volume.h
+++ b/libavfilter/af_volume.h
@@ -48,6 +48,7 @@ typedef struct VolumeContext {
     void (*scale_samples)(uint8_t *dst, const uint8_t *src, int
nb_samples,
                           int volume);
     int samples_align;
+    char *metadata;
 } VolumeContext;
void ff_volume_init_x86(VolumeContext *vol);
diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c
index 88d37e8..f4ce6d9 100644
--- a/libavfilter/f_ebur128.c
+++ b/libavfilter/f_ebur128.c
@@ -410,7 +410,7 @@ static av_cold int init(AVFilterContext *ctx)
if (ebur128->loglevel != AV_LOG_INFO &&
         ebur128->loglevel != AV_LOG_VERBOSE) {
-        if (ebur128->do_video || ebur128->metadata)
+        if (ebur128->do_video)
             ebur128->loglevel = AV_LOG_VERBOSE;
         else
             ebur128->loglevel = AV_LOG_INFO;
@@ -689,7 +689,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
*insamples)
                 SET_META("LRA.high", ebur128->lra_high);
             }
-            av_log(ctx, ebur128->loglevel, "t: %-10s " LOG_FMT "\n",
+            av_log(ctx, ebur128->metadata || !ebur128->do_video ?
AV_LOG_VERBOSE : ebur128->loglevel, "t: %-10s " LOG_FMT "\n",
                    av_ts2timestr(pts, &outlink->time_base),
                    loudness_400, loudness_3000,
                    ebur128->integrated_loudness, ebur128->loudness_range);
diff --git a/libavfilter/x86/af_volume_init.c
b/libavfilter/x86/af_volume_init.c
index 81d605f..fab5a03 100644
--- a/libavfilter/x86/af_volume_init.c
+++ b/libavfilter/x86/af_volume_init.c
@@ -39,7 +39,7 @@ av_cold void ff_volume_init_x86(VolumeContext *vol)
     enum AVSampleFormat sample_fmt =
av_get_packed_sample_fmt(vol->sample_fmt);
if (sample_fmt == AV_SAMPLE_FMT_S16) {
-        if (EXTERNAL_SSE2(mm_flags) && vol->volume_i < 32768) {
+        if (EXTERNAL_SSE2(mm_flags) && vol->volume_i < 32768 &&
Why? This is suboptimal.
...
!vol->metadata) {
             vol->scale_samples = ff_scale_samples_s16_sse2;
             vol->samples_align = 8;
         }
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Re: [FFmpeg-devel] One pass volume normalization (ebur128)

Paul B Mahol