Merge master.

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

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index 7bc933fd01b2536da1b8d7f0e70c6a9264d6a6f0..e414436e8b39960c6c5f78ca16188c0781bc0407 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) 2013-2014 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
@@ -19,10 +19,12 @@
 
 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"
 
@@ -36,24 +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);
 }
@@ -63,11 +60,9 @@ Resampler::~Resampler ()
        swr_free (&_swr_context);
 }
 
-pair<shared_ptr<const AudioBuffers>, AudioContent::Frame>
-Resampler::run (shared_ptr<const AudioBuffers> in, AudioContent::Frame frame)
+shared_ptr<const AudioBuffers>
+Resampler::run (shared_ptr<const AudioBuffers> in)
 {
-       AudioContent::Frame const resamp_time = swr_next_pts (_swr_context, frame * _out_rate) / _in_rate;
-               
        /* Compute the resampled frames count and add 32 for luck */
        int const max_resampled_frames = ceil ((double) in->frames() * _out_rate / _in_rate) + 32;
        shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, max_resampled_frames));
@@ -77,11 +72,13 @@ Resampler::run (shared_ptr<const AudioBuffers> in, AudioContent::Frame frame)
                );
        
        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);
-       return make_pair (resampled, resamp_time);
+       return resampled;
 }      
 
 shared_ptr<const AudioBuffers>
@@ -92,7 +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) {