X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fresampler.cc;h=f54f4ec6abd3ab16c16ac48de8a408f93d4f254a;hb=e526999a0252193ee8e42ac841c7494bea44ecfb;hp=7bc933fd01b2536da1b8d7f0e70c6a9264d6a6f0;hpb=373f010a7f04add1f49169cbaa60cb7ae5f508d4;p=dcpomatic.git

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index 7bc933fd0..f54f4ec6a 100644
--- 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
@@ -19,10 +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"
 
@@ -36,25 +38,22 @@ 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);
+
+	av_opt_set (_swr_context, "resampler", "soxr", 0);
+
 	swr_init (_swr_context);
 }
 
@@ -63,11 +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));
@@ -75,14 +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) {
-		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>
 Resampler::flush ()
@@ -92,13 +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;
 		}