/*
- 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
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)
, _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.
- */
-
- cout << "resamp for " << _channels << " " << _in_rate << " " << _out_rate << "\n";
-
- _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);
}
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 resampled;
-}
+}
shared_ptr<const AudioBuffers>
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;
}