/*
 * 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) noexcept
{
	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) noexcept
{
	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;
}
