Use soxr resampler.

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

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index e6b1623d9cbeb3afbe7df32a758323d9980b36b9..f54f4ec6abd3ab16c16ac48de8a408f93d4f254a 100644 (file)
--- a/src/lib/resampler.cc
+++ b/src/lib/resampler.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2013-2014 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
@@ -20,7 +20,7 @@
 extern "C" {
 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
-}      
+}
 #include "resampler.h"
 #include "audio_buffers.h"
 #include "exceptions.h"
@@ -51,7 +51,9 @@ Resampler::Resampler (int in, int out, int channels)
        /* Sample rates */
        av_opt_set_int (_swr_context, "isr", _in_rate, 0);
        av_opt_set_int (_swr_context, "osr", _out_rate, 0);
-       
+
+       av_opt_set (_swr_context, "resampler", "soxr", 0);
+
        swr_init (_swr_context);
 }
 
@@ -60,11 +62,9 @@ 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));
@@ -72,16 +72,16 @@ Resampler::run (shared_ptr<const AudioBuffers> in, AudioContent::Frame frame)
        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 make_pair (resampled, resamp_time);
-}      
+       return resampled;
+}
 
 shared_ptr<const AudioBuffers>
 Resampler::flush ()
@@ -91,13 +91,13 @@ 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;
                }