Files @ r23483:3733e6b8ff17
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Location: cpp/openttd-patchpack/source/src/mixer.cpp

Patric Stout
Remove: ENABLE_NETWORK switch

This switch has been a pain for years. Often disabling broke
compilation, as no developer compiles OpenTTD without, neither do
any of our official binaries.

Additionaly, it has grown so hugely in our codebase, that it
clearly shows that the current solution was a poor one. 350+
instances of "#ifdef ENABLE_NETWORK" were in the code, of which
only ~30 in the networking code itself. The rest were all around
the code to do the right thing, from GUI to NewGRF.

A more proper solution would be to stub all the functions, and
make sure the rest of the code can simply assume network is
available. This was also partially done, and most variables were
correct if networking was disabled. Despite that, often the #ifdefs
were still used.

With the recent removal of DOS, there is also no platform anymore
which we support where networking isn't working out-of-the-box.

All in all, it is time to remove the ENABLE_NETWORK switch. No
replacement is planned, but if you feel we really need this option,
we welcome any Pull Request which implements this in a way that
doesn't crawl through the code like this diff shows we used to.
/* $Id$ */

/*
 * This file is part of OpenTTD.
 * OpenTTD 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, version 2.
 * OpenTTD 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 OpenTTD. If not, see <http://www.gnu.org/licenses/>.
 */

/** @file mixer.cpp Mixing of sound samples. */

#include "stdafx.h"
#include <math.h>
#include "core/math_func.hpp"
#include "framerate_type.h"

#include "safeguards.h"
#include "mixer.h"

struct MixerChannel {
	bool active;

	/* pointer to allocated buffer memory */
	int8 *memory;

	/* current position in memory */
	uint32 pos;
	uint32 frac_pos;
	uint32 frac_speed;
	uint32 samples_left;

	/* Mixing volume */
	int volume_left;
	int volume_right;

	bool is16bit;
};

static MixerChannel _channels[8];
static uint32 _play_rate = 11025;
static uint32 _max_size = UINT_MAX;
static MxStreamCallback _music_stream = NULL;

/**
 * The theoretical maximum volume for a single sound sample. Multiple sound
 * samples should not exceed this limit as it will sound too loud. It also
 * stops overflowing when too many sounds are played at the same time, which
 * causes an even worse sound quality.
 */
static const int MAX_VOLUME = 128 * 128;

/**
 * Perform the rate conversion between the input and output.
 * @param b the buffer to read the data from
 * @param frac_pos the position from the begin of the buffer till the next element
 * @tparam T the size of the buffer (8 or 16 bits)
 * @return the converted value.
 */
template <typename T>
static int RateConversion(T *b, int frac_pos)
{
	return ((b[0] * ((1 << 16) - frac_pos)) + (b[1] * frac_pos)) >> 16;
}

static void mix_int16(MixerChannel *sc, int16 *buffer, uint samples)
{
	if (samples > sc->samples_left) samples = sc->samples_left;
	sc->samples_left -= samples;
	assert(samples > 0);

	const int16 *b = (const int16 *)sc->memory + sc->pos;
	uint32 frac_pos = sc->frac_pos;
	uint32 frac_speed = sc->frac_speed;
	int volume_left = sc->volume_left;
	int volume_right = sc->volume_right;

	if (frac_speed == 0x10000) {
		/* Special case when frac_speed is 0x10000 */
		do {
			buffer[0] = Clamp(buffer[0] + (*b * volume_left  >> 16), -MAX_VOLUME, MAX_VOLUME);
			buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
			b++;
			buffer += 2;
		} while (--samples > 0);
	} else {
		do {
			int data = RateConversion(b, frac_pos);
			buffer[0] = Clamp(buffer[0] + (data * volume_left  >> 16), -MAX_VOLUME, MAX_VOLUME);
			buffer[1] = Clamp(buffer[1] + (data * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
			buffer += 2;
			frac_pos += frac_speed;
			b += frac_pos >> 16;
			frac_pos &= 0xffff;
		} while (--samples > 0);
	}

	sc->frac_pos = frac_pos;
	sc->pos = b - (const int16 *)sc->memory;
}

static void mix_int8_to_int16(MixerChannel *sc, int16 *buffer, uint samples)
{
	if (samples > sc->samples_left) samples = sc->samples_left;
	sc->samples_left -= samples;
	assert(samples > 0);

	const int8 *b = sc->memory + sc->pos;
	uint32 frac_pos = sc->frac_pos;
	uint32 frac_speed = sc->frac_speed;
	int volume_left = sc->volume_left;
	int volume_right = sc->volume_right;

	if (frac_speed == 0x10000) {
		/* Special case when frac_speed is 0x10000 */
		do {
			buffer[0] = Clamp(buffer[0] + (*b * volume_left  >> 8), -MAX_VOLUME, MAX_VOLUME);
			buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 8), -MAX_VOLUME, MAX_VOLUME);
			b++;
			buffer += 2;
		} while (--samples > 0);
	} else {
		do {
			int data = RateConversion(b, frac_pos);
			buffer[0] = Clamp(buffer[0] + (data * volume_left  >> 8), -MAX_VOLUME, MAX_VOLUME);
			buffer[1] = Clamp(buffer[1] + (data * volume_right >> 8), -MAX_VOLUME, MAX_VOLUME);
			buffer += 2;
			frac_pos += frac_speed;
			b += frac_pos >> 16;
			frac_pos &= 0xffff;
		} while (--samples > 0);
	}

	sc->frac_pos = frac_pos;
	sc->pos = b - sc->memory;
}

static void MxCloseChannel(MixerChannel *mc)
{
	mc->active = false;
}

void MxMixSamples(void *buffer, uint samples)
{
	PerformanceMeasurer framerate(PFE_SOUND);
	static uint last_samples = 0;
	if (samples != last_samples) {
		framerate.SetExpectedRate((double)_play_rate / samples);
		last_samples = samples;
	}

	MixerChannel *mc;

	/* Clear the buffer */
	memset(buffer, 0, sizeof(int16) * 2 * samples);

	/* Fetch music if a sampled stream is available */
	if (_music_stream) _music_stream((int16*)buffer, samples);

	/* Mix each channel */
	for (mc = _channels; mc != endof(_channels); mc++) {
		if (mc->active) {
			if (mc->is16bit) {
				mix_int16(mc, (int16*)buffer, samples);
			} else {
				mix_int8_to_int16(mc, (int16*)buffer, samples);
			}
			if (mc->samples_left == 0) MxCloseChannel(mc);
		}
	}
}

MixerChannel *MxAllocateChannel()
{
	MixerChannel *mc;
	for (mc = _channels; mc != endof(_channels); mc++) {
		if (!mc->active) {
			free(mc->memory);
			mc->memory = NULL;
			return mc;
		}
	}
	return NULL;
}

void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate, bool is16bit)
{
	mc->memory = mem;
	mc->frac_pos = 0;
	mc->pos = 0;

	mc->frac_speed = (rate << 16) / _play_rate;

	if (is16bit) size /= 2;

	/* adjust the magnitude to prevent overflow */
	while (size >= _max_size) {
		size >>= 1;
		rate = (rate >> 1) + 1;
	}

	mc->samples_left = (uint)size * _play_rate / rate;
	mc->is16bit = is16bit;
}

/**
 * Set volume and pan parameters for a sound.
 * @param mc     MixerChannel to set
 * @param volume Volume level for sound, range is 0..16384
 * @param pan    Pan position for sound, range is 0..1
 */
void MxSetChannelVolume(MixerChannel *mc, uint volume, float pan)
{
	/* Use sinusoidal pan to maintain overall sound power level regardless
	 * of position. */
	mc->volume_left = (uint)(sin((1.0 - pan) * M_PI / 2.0) * volume);
	mc->volume_right = (uint)(sin(pan * M_PI / 2.0) * volume);
}


void MxActivateChannel(MixerChannel *mc)
{
	mc->active = true;
}

/**
 * Set source of PCM music
 * @param music_callback Function that will be called to fill sample buffers with music data.
 * @return Sample rate of mixer, which the buffers supplied to the callback must be rendered at.
 */
uint32 MxSetMusicSource(MxStreamCallback music_callback)
{
	_music_stream = music_callback;
	return _play_rate;
}


bool MxInitialize(uint rate)
{
	_play_rate = rate;
	_max_size  = UINT_MAX / _play_rate;
	_music_stream = NULL; /* rate may have changed, any music source is now invalid */
	return true;
}