File audio_resample-issue#2126.patch of Package handbrake

File audio_resample-issue#2126.patch of Package handbrake (Revision 4ee34d73bb5af65f8ffeb367f407e794)

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commit 4ba75f97f31b0e6f64ebd4684d5f2aeeba494629
Author: John Stebbins <jstebbins.hb@gmail.com>
Date:   Thu May 30 08:45:13 2019 -0700

audio_resample: set input and output samplerate
    
    Fixes https://github.com/HandBrake/HandBrake/issues/2124

diff -rup HandBrake-1.2.2.orig/libhb/audio_resample.c HandBrake-1.2.2/libhb/audio_resample.c
--- HandBrake-1.2.2.orig/libhb/audio_resample.c	2019-02-22 17:23:51.000000000 +0100
+++ HandBrake-1.2.2/libhb/audio_resample.c	2019-10-14 14:32:45.006416242 +0200
@@ -12,6 +12,7 @@
 #include "audio_resample.h"
 
 hb_audio_resample_t* hb_audio_resample_init(enum AVSampleFormat sample_fmt,
+                                            int sample_rate,
                                             int hb_amixdown, int normalize_mix)
 {
     hb_audio_resample_t *resample = calloc(1, sizeof(hb_audio_resample_t));
@@ -56,6 +57,7 @@ hb_audio_resample_t* hb_audio_resample_i
     resample->out.channel_layout      = channel_layout;
     resample->out.matrix_encoding     = matrix_encoding;
     resample->out.sample_fmt          = sample_fmt;
+    resample->out.sample_rate         = sample_rate;
     if (normalize_mix)
     {
         resample->out.maxval = 1.0;
@@ -68,6 +70,7 @@ hb_audio_resample_t* hb_audio_resample_i
 
     // set default input characteristics
     resample->in.sample_fmt         = resample->out.sample_fmt;
+    resample->in.sample_rate        = resample->out.sample_rate;
     resample->in.channel_layout     = resample->out.channel_layout;
     resample->in.lfe_mix_level      = HB_MIXLEV_ZERO;
     resample->in.center_mix_level   = HB_MIXLEV_DEFAULT;
@@ -145,6 +148,15 @@ void hb_audio_resample_set_sample_fmt(hb
     }
 }
 
+void hb_audio_resample_set_sample_rate(hb_audio_resample_t *resample,
+                                       int sample_rate)
+{
+    if (resample != NULL)
+    {
+        resample->in.sample_rate = sample_rate;
+    }
+}
+
 int hb_audio_resample_update(hb_audio_resample_t *resample)
 {
     if (resample == NULL)
@@ -157,11 +169,13 @@ int hb_audio_resample_update(hb_audio_re
 
     resample->resample_needed =
         (resample->out.sample_fmt != resample->in.sample_fmt ||
+         resample->out.sample_rate != resample->in.sample_rate ||
          resample->out.channel_layout != resample->in.channel_layout);
 
     resample_changed =
         (resample->resample_needed &&
          (resample->resample.sample_fmt != resample->in.sample_fmt ||
+          resample->resample.sample_rate != resample->in.sample_rate ||
           resample->resample.channel_layout != resample->in.channel_layout ||
           resample->resample.lfe_mix_level != resample->in.lfe_mix_level ||
           resample->resample.center_mix_level != resample->in.center_mix_level ||
@@ -181,6 +195,8 @@ int hb_audio_resample_update(hb_audio_re
 
             av_opt_set_int(resample->swresample, "out_sample_fmt",
                            resample->out.sample_fmt, 0);
+            av_opt_set_int(resample->swresample, "out_sample_rate",
+                           resample->out.sample_rate, 0);
             av_opt_set_int(resample->swresample, "out_channel_layout",
                            resample->out.channel_layout, 0);
             av_opt_set_int(resample->swresample, "matrix_encoding",
@@ -191,6 +207,8 @@ int hb_audio_resample_update(hb_audio_re
 
         av_opt_set_int(resample->swresample, "in_sample_fmt",
                        resample->in.sample_fmt, 0);
+        av_opt_set_int(resample->swresample, "in_sample_rate",
+                       resample->in.sample_rate, 0);
         av_opt_set_int(resample->swresample, "in_channel_layout",
                        resample->in.channel_layout, 0);
         av_opt_set_double(resample->swresample, "lfe_mix_level",
@@ -212,6 +230,7 @@ int hb_audio_resample_update(hb_audio_re
         }
 
         resample->resample.sample_fmt         = resample->in.sample_fmt;
+        resample->resample.sample_rate        = resample->in.sample_rate;
         resample->resample.channel_layout     = resample->in.channel_layout;
         resample->resample.channels           =
             av_get_channel_layout_nb_channels(resample->in.channel_layout);
@@ -255,11 +274,13 @@ hb_buffer_t* hb_audio_resample(hb_audio_
 
     if (resample->resample_needed)
     {
-        out_size = av_samples_get_buffer_size(NULL,
-                                              resample->out.channels, nsamples,
+        out_samples = (nsamples + 1) * resample->out.sample_rate /
+                                       resample->in.sample_rate;
+        out_size = av_samples_get_buffer_size(NULL, resample->out.channels,
+                                              out_samples,
                                               resample->out.sample_fmt, 0);
         out = hb_buffer_init(out_size);
-        out_samples = swr_convert(resample->swresample, &out->data, nsamples,
+        out_samples = swr_convert(resample->swresample, &out->data, out_samples,
                                                         samples,    nsamples);
 
         if (out_samples <= 0)
diff -rup HandBrake-1.2.2.orig/libhb/audio_resample.h HandBrake-1.2.2/libhb/audio_resample.h
--- HandBrake-1.2.2.orig/libhb/audio_resample.h	2019-02-22 17:23:51.000000000 +0100
+++ HandBrake-1.2.2/libhb/audio_resample.h	2019-10-14 14:32:45.006416242 +0200
@@ -37,6 +37,7 @@ typedef struct
 
     struct
     {
+        int sample_rate;
         uint64_t channel_layout;
         double lfe_mix_level;
         double center_mix_level;
@@ -46,6 +47,7 @@ typedef struct
 
     struct
     {
+        int sample_rate;
         int channels;
         uint64_t channel_layout;
         double lfe_mix_level;
@@ -56,6 +58,7 @@ typedef struct
 
     struct
     {
+        int sample_rate;
         int channels;
         int sample_size;
         uint64_t channel_layout;
@@ -72,6 +75,7 @@ typedef struct
  * as the output characteristics (no conversion needed).
  */
 hb_audio_resample_t* hb_audio_resample_init(enum AVSampleFormat sample_fmt,
+                                            int sample_rate,
                                             int hb_amixdown, int normalize_mix);
 
 /* The following functions set the audio input characteristics.
@@ -91,6 +95,9 @@ void                 hb_audio_resample_s
 void                 hb_audio_resample_set_sample_fmt(hb_audio_resample_t *resample,
                                                       enum AVSampleFormat sample_fmt);
 
+void                 hb_audio_resample_set_sample_rate(hb_audio_resample_t *resample,
+                                                       int sample_rate);
+
 /* Update an hb_audio_resample_t.
  *
  * Must be called after using any of the above functions.
diff -rup HandBrake-1.2.2.orig/libhb/decavcodec.c HandBrake-1.2.2/libhb/decavcodec.c
--- HandBrake-1.2.2.orig/libhb/decavcodec.c	2019-02-22 17:23:51.000000000 +0100
+++ HandBrake-1.2.2/libhb/decavcodec.c	2019-10-14 14:32:45.006416242 +0200
@@ -204,8 +204,13 @@ static int decavcodecaInit( hb_work_obje
     /* Downmixing & sample_fmt conversion */
     if (!(w->audio->config.out.codec & HB_ACODEC_PASS_FLAG))
     {
+        // Currently, samplerate conversion is performed in sync.c
+        // So set output samplerate to input samplerate
+        // This should someday get reworked to be part of an audio
+        // filter pipleine.
         pv->resample =
             hb_audio_resample_init(AV_SAMPLE_FMT_FLT,
+                                   w->audio->config.in.samplerate,
                                    w->audio->config.out.mixdown,
                                    w->audio->config.out.normalize_mix_level);
         if (pv->resample == NULL)
@@ -2308,6 +2313,8 @@ static void decodeAudio(hb_work_private_
             hb_audio_resample_set_channel_layout(pv->resample, channel_layout);
             hb_audio_resample_set_sample_fmt(pv->resample,
                                              pv->frame->format);
+            hb_audio_resample_set_sample_rate(pv->resample,
+                                             pv->frame->sample_rate);
             if (hb_audio_resample_update(pv->resample))
             {
                 hb_log("decavcodec: hb_audio_resample_update() failed");
Only in HandBrake-1.2.2/libhb: decavcodec.c.orig
diff -rup HandBrake-1.2.2.orig/libhb/declpcm.c HandBrake-1.2.2/libhb/declpcm.c
--- HandBrake-1.2.2.orig/libhb/declpcm.c	2019-02-22 17:23:51.000000000 +0100
+++ HandBrake-1.2.2/libhb/declpcm.c	2019-10-14 14:32:45.006416242 +0200
@@ -166,8 +166,12 @@ static int declpcmInit( hb_work_object_t
     pv->job = job;
 
     pv->next_pts = (int64_t)AV_NOPTS_VALUE;
+    // Currently, samplerate conversion is performed in sync.c
+    // So set output samplerate to input samplerate
+    // This should someday get reworked to be part of an audio filter pipleine.
     pv->resample =
         hb_audio_resample_init(AV_SAMPLE_FMT_FLT,
+                               w->audio->config.in.samplerate,
                                w->audio->config.out.mixdown,
                                w->audio->config.out.normalize_mix_level);
     if (pv->resample == NULL)
@@ -333,6 +337,8 @@ static hb_buffer_t *Decode( hb_work_obje
 
     hb_audio_resample_set_channel_layout(pv->resample,
                                          hdr2layout[pv->nchannels - 1]);
+    hb_audio_resample_set_sample_rate(pv->resample,
+                                      pv->samplerate);
     if (hb_audio_resample_update(pv->resample))
     {
         hb_log("declpcm: hb_audio_resample_update() failed");
diff -rup HandBrake-1.2.2.orig/libhb/encavcodecaudio.c HandBrake-1.2.2/libhb/encavcodecaudio.c
--- HandBrake-1.2.2.orig/libhb/encavcodecaudio.c	2019-02-22 17:23:51.000000000 +0100
+++ HandBrake-1.2.2/libhb/encavcodecaudio.c	2019-10-14 14:32:45.006416242 +0200
@@ -249,13 +249,13 @@ static int encavcodecaInit(hb_work_objec
                        context->channel_layout, 0);
         av_opt_set_int(pv->swresample, "out_channel_layout",
                        context->channel_layout, 0);
+        av_opt_set_int(pv->swresample, "in_sample_rate",
+                       context->sample_rate, 0);
+        av_opt_set_int(pv->swresample, "out_sample_rate",
+                       context->sample_rate, 0);
         if (hb_audio_dither_is_supported(audio->config.out.codec))
         {
             // dithering needs the sample rate
-            av_opt_set_int(pv->swresample, "in_sample_rate",
-                           context->sample_rate, 0);
-            av_opt_set_int(pv->swresample, "out_sample_rate",
-                           context->sample_rate, 0);
             av_opt_set_int(pv->swresample, "dither_method",
                            audio->config.out.dither_method, 0);
         }