Packman Build Service PMBS

Changes of Revision 31

handbrake.changes Changed

handbrake.spec Changed

audio_resample-issue#2126.patch Added

@@ -0,0 +1,226 @@
+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);
+         }