No-op: remove all trailing whitespace.

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

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index 00121384dce7617ab59f0ee85ea0373a7ab0c440..2538f7dca57080d718954fdb6f216753b78ae7c2 100644 (file)
--- a/src/lib/resampler.cc
+++ b/src/lib/resampler.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2013 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2013-2015 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
@@ -19,7 +19,8 @@
 
 extern "C" {
 #include "libavutil/channel_layout.h"
-}      
+#include "libavutil/opt.h"
+}
 #include "resampler.h"
 #include "audio_buffers.h"
 #include "exceptions.h"
@@ -37,25 +38,20 @@ 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);
 }
 
@@ -74,16 +70,16 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
        int const resampled_frames = swr_convert (
                _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) in->data(), in->frames()
                );
-       
+
        if (resampled_frames < 0) {
                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);
        return resampled;
-}      
+}
 
 shared_ptr<const AudioBuffers>
 Resampler::flush ()
@@ -93,13 +89,13 @@ 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) {
                        throw EncodeError (_("could not run sample-rate converter"));
                }
-               
+
                if (frames == 0) {
                        break;
                }