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/*
* Copyright 2003-2017 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "AudioFormat.hxx"
#include "util/StringBuffer.hxx"
#include <assert.h>
#include <stdio.h>
void
AudioFormat::ApplyMask(AudioFormat mask)
{
assert(IsValid());
assert(mask.IsMaskValid());
if (mask.sample_rate != 0)
sample_rate = mask.sample_rate;
if (mask.format != SampleFormat::UNDEFINED)
format = mask.format;
if (mask.channels != 0)
channels = mask.channels;
assert(IsValid());
}
StringBuffer<24>
ToString(const AudioFormat af)
{
StringBuffer<24> buffer;
if (af.format == SampleFormat::DSD && af.sample_rate > 0 &&
af.sample_rate % 44100 == 0) {
/* use shortcuts such as "dsd64" which implies the
sample rate */
snprintf(buffer.data(), buffer.capacity(), "dsd%u:%u",
af.sample_rate * 8 / 44100,
af.channels);
return buffer;
}
snprintf(buffer.data(), buffer.capacity(), "%u:%s:%u",
af.sample_rate, sample_format_to_string(af.format),
af.channels);
return buffer;
}
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