Merge master.

[dcpomatic.git] / src / lib / resampler.cc

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index ebb507cb124fd8ca68e93b813480aec581c209d2..e414436e8b39960c6c5f78ca16188c0781bc0407 100644 (file)
--- a/src/lib/resampler.cc
+++ b/src/lib/resampler.cc
@@ -1,12 +1,36 @@
+/*
+    Copyright (C) 2013-2014 Carl Hetherington <cth@carlh.net>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+*/
+
 extern "C" {
 #include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
 }      
 #include "resampler.h"
 #include "audio_buffers.h"
 #include "exceptions.h"
+#include "compose.hpp"
 
 #include "i18n.h"
 
+using std::cout;
+using std::pair;
+using std::make_pair;
 using boost::shared_ptr;
 
 Resampler::Resampler (int in, int out, int channels)
@@ -14,24 +38,19 @@ Resampler::Resampler (int in, int out, int channels)
        , _out_rate (out)
        , _channels (channels)
 {
-       /* We will be using planar float data when we call the
-          resampler.  As far as I can see, the audio channel
-          layout is not necessary for our purposes; it seems
-          only to be used get the number of channels and
-          decide if rematrixing is needed.  It won't be, since
-          input and output layouts are the same.
-       */
-
-       _swr_context = swr_alloc_set_opts (
-               0,
-               av_get_default_channel_layout (_channels),
-               AV_SAMPLE_FMT_FLTP,
-               _out_rate,
-               av_get_default_channel_layout (_channels),
-               AV_SAMPLE_FMT_FLTP,
-               _in_rate,
-               0, 0
-               );
+       _swr_context = swr_alloc ();
+
+       /* Sample formats */
+       av_opt_set_int (_swr_context, "isf", AV_SAMPLE_FMT_FLTP, 0);
+       av_opt_set_int (_swr_context, "osf", AV_SAMPLE_FMT_FLTP, 0);
+
+       /* Channel counts */
+       av_opt_set_int (_swr_context, "ich", _channels, 0);
+       av_opt_set_int (_swr_context, "och", _channels, 0);
+
+       /* Sample rates */
+       av_opt_set_int (_swr_context, "isr", _in_rate, 0);
+       av_opt_set_int (_swr_context, "osr", _out_rate, 0);
        
        swr_init (_swr_context);
 }
@@ -53,7 +72,9 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
                );
        
        if (resampled_frames < 0) {
-               throw EncodeError (_("could not run sample-rate converter"));
+               char buf[256];
+               av_strerror (resampled_frames, buf, sizeof(buf));
+               throw EncodeError (String::compose (_("could not run sample-rate converter for %1 samples (%2) (%3)"), in->frames(), resampled_frames, buf));
        }
        
        resampled->set_frames (resampled_frames);
@@ -68,7 +89,7 @@ Resampler::flush ()
        int64_t const pass_size = 256;
        shared_ptr<AudioBuffers> pass (new AudioBuffers (_channels, 256));
 
-       while (1) {
+       while (true) {
                int const frames = swr_convert (_swr_context, (uint8_t **) pass->data(), pass_size, 0, 0);
                
                if (frames < 0) {