mirror of
https://github.com/JGRennison/OpenTTD-patches.git
synced 2024-11-11 13:10:45 +00:00
270 lines
8.0 KiB
C++
270 lines
8.0 KiB
C++
/*
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* This file is part of OpenTTD.
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* 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.
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* 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.
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* 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/>.
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*/
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/** @file mixer.cpp Mixing of sound samples. */
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#include "stdafx.h"
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#include <math.h>
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#include <mutex>
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#include <atomic>
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#include "core/math_func.hpp"
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#include "framerate_type.h"
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#include "settings_type.h"
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#include "safeguards.h"
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#include "mixer.h"
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#include <mutex>
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#if defined(__MINGW32__)
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#include "3rdparty/mingw-std-threads/mingw.mutex.h"
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#endif
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struct MixerChannel {
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/* pointer to allocated buffer memory */
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int8 *memory;
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/* current position in memory */
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uint32 pos;
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uint32 frac_pos;
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uint32 frac_speed;
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uint32 samples_left;
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/* Mixing volume */
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int volume_left;
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int volume_right;
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bool is16bit;
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};
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static std::atomic<uint8> _active_channels;
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static MixerChannel _channels[8];
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static uint32 _play_rate = 11025;
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static uint32 _max_size = UINT_MAX;
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static MxStreamCallback _music_stream = nullptr;
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static std::mutex _music_stream_mutex;
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static std::atomic<uint8> _effect_vol;
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/**
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* The theoretical maximum volume for a single sound sample. Multiple sound
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* samples should not exceed this limit as it will sound too loud. It also
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* stops overflowing when too many sounds are played at the same time, which
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* causes an even worse sound quality.
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*/
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static const int MAX_VOLUME = 32767;
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/**
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* Perform the rate conversion between the input and output.
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* @param b the buffer to read the data from
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* @param frac_pos the position from the begin of the buffer till the next element
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* @tparam T the size of the buffer (8 or 16 bits)
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* @return the converted value.
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*/
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template <typename T>
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static int RateConversion(T *b, int frac_pos)
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{
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return ((b[0] * ((1 << 16) - frac_pos)) + (b[1] * frac_pos)) >> 16;
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}
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static void mix_int16(MixerChannel *sc, int16 *buffer, uint samples, uint8 effect_vol)
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{
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if (samples > sc->samples_left) samples = sc->samples_left;
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sc->samples_left -= samples;
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assert(samples > 0);
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const int16 *b = (const int16 *)sc->memory + sc->pos;
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uint32 frac_pos = sc->frac_pos;
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uint32 frac_speed = sc->frac_speed;
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int volume_left = sc->volume_left * effect_vol / 255;
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int volume_right = sc->volume_right * effect_vol / 255;
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if (frac_speed == 0x10000) {
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/* Special case when frac_speed is 0x10000 */
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do {
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buffer[0] = Clamp(buffer[0] + (*b * volume_left >> 16), -MAX_VOLUME, MAX_VOLUME);
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buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
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b++;
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buffer += 2;
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} while (--samples > 0);
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} else {
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do {
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int data = RateConversion(b, frac_pos);
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buffer[0] = Clamp(buffer[0] + (data * volume_left >> 16), -MAX_VOLUME, MAX_VOLUME);
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buffer[1] = Clamp(buffer[1] + (data * volume_right >> 16), -MAX_VOLUME, MAX_VOLUME);
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buffer += 2;
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frac_pos += frac_speed;
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b += frac_pos >> 16;
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frac_pos &= 0xffff;
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} while (--samples > 0);
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}
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sc->frac_pos = frac_pos;
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sc->pos = b - (const int16 *)sc->memory;
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}
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static void mix_int8_to_int16(MixerChannel *sc, int16 *buffer, uint samples, uint8 effect_vol)
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{
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if (samples > sc->samples_left) samples = sc->samples_left;
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sc->samples_left -= samples;
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assert(samples > 0);
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const int8 *b = sc->memory + sc->pos;
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uint32 frac_pos = sc->frac_pos;
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uint32 frac_speed = sc->frac_speed;
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int volume_left = sc->volume_left * effect_vol / 255;
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int volume_right = sc->volume_right * effect_vol / 255;
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if (frac_speed == 0x10000) {
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/* Special case when frac_speed is 0x10000 */
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do {
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buffer[0] = Clamp(buffer[0] + (*b * volume_left >> 8), -MAX_VOLUME, MAX_VOLUME);
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buffer[1] = Clamp(buffer[1] + (*b * volume_right >> 8), -MAX_VOLUME, MAX_VOLUME);
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b++;
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buffer += 2;
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} while (--samples > 0);
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} else {
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do {
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int data = RateConversion(b, frac_pos);
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buffer[0] = Clamp(buffer[0] + (data * volume_left >> 8), -MAX_VOLUME, MAX_VOLUME);
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buffer[1] = Clamp(buffer[1] + (data * volume_right >> 8), -MAX_VOLUME, MAX_VOLUME);
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buffer += 2;
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frac_pos += frac_speed;
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b += frac_pos >> 16;
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frac_pos &= 0xffff;
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} while (--samples > 0);
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}
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sc->frac_pos = frac_pos;
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sc->pos = b - sc->memory;
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}
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static void MxCloseChannel(uint8 channel_index)
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{
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_active_channels.fetch_and(~(1 << channel_index), std::memory_order_release);
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}
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void MxMixSamples(void *buffer, uint samples)
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{
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PerformanceMeasurer framerate(PFE_SOUND);
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static uint last_samples = 0;
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if (samples != last_samples) {
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framerate.SetExpectedRate((double)_play_rate / samples);
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last_samples = samples;
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}
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/* Clear the buffer */
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memset(buffer, 0, sizeof(int16) * 2 * samples);
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{
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std::lock_guard<std::mutex> lock{ _music_stream_mutex };
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/* Fetch music if a sampled stream is available */
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if (_music_stream) _music_stream((int16*)buffer, samples);
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}
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/* Apply simple x^3 scaling to master effect volume. This increases the
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* perceived difference in loudness to better match expectations. effect_vol
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* is expected to be in the range 0-127 hence the division by 127 * 127 to
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* get back into range. */
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uint8 effect_vol_setting = _effect_vol.load(std::memory_order_relaxed);
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uint8 effect_vol = (effect_vol_setting *
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effect_vol_setting *
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effect_vol_setting) / (127 * 127);
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/* Mix each channel */
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uint8 active = _active_channels.load(std::memory_order_acquire);
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for (uint8 idx : SetBitIterator(active)) {
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MixerChannel *mc = &_channels[idx];
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if (mc->is16bit) {
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mix_int16(mc, (int16*)buffer, samples, effect_vol);
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} else {
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mix_int8_to_int16(mc, (int16*)buffer, samples, effect_vol);
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}
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if (mc->samples_left == 0) MxCloseChannel(idx);
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}
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}
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MixerChannel *MxAllocateChannel()
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{
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uint8 currently_active = _active_channels.load(std::memory_order_acquire);
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uint8 available = ~currently_active;
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if (available == 0) return nullptr;
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uint8 channel_index = FindFirstBit(available);
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MixerChannel *mc = &_channels[channel_index];
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free(mc->memory);
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mc->memory = nullptr;
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return mc;
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}
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void MxSetChannelRawSrc(MixerChannel *mc, int8 *mem, size_t size, uint rate, bool is16bit)
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{
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mc->memory = mem;
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mc->frac_pos = 0;
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mc->pos = 0;
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mc->frac_speed = (rate << 16) / _play_rate;
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if (is16bit) size /= 2;
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/* adjust the magnitude to prevent overflow */
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while (size >= _max_size) {
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size >>= 1;
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rate = (rate >> 1) + 1;
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}
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mc->samples_left = (uint)size * _play_rate / rate;
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mc->is16bit = is16bit;
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}
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/**
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* Set volume and pan parameters for a sound.
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* @param mc MixerChannel to set
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* @param volume Volume level for sound, range is 0..16384
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* @param pan Pan position for sound, range is 0..1
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*/
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void MxSetChannelVolume(MixerChannel *mc, uint volume, float pan)
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{
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/* Use sinusoidal pan to maintain overall sound power level regardless
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* of position. */
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mc->volume_left = (uint)(sin((1.0 - pan) * M_PI / 2.0) * volume);
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mc->volume_right = (uint)(sin(pan * M_PI / 2.0) * volume);
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}
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void MxActivateChannel(MixerChannel *mc)
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{
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uint8 channel_index = mc - _channels;
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_active_channels.fetch_or((1 << channel_index), std::memory_order_release);
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}
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/**
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* Set source of PCM music
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* @param music_callback Function that will be called to fill sample buffers with music data.
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* @return Sample rate of mixer, which the buffers supplied to the callback must be rendered at.
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*/
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uint32 MxSetMusicSource(MxStreamCallback music_callback)
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{
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std::lock_guard<std::mutex> lock{ _music_stream_mutex };
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_music_stream = music_callback;
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return _play_rate;
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}
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bool MxInitialize(uint rate)
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{
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std::lock_guard<std::mutex> lock{ _music_stream_mutex };
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_play_rate = rate;
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_max_size = UINT_MAX / _play_rate;
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_music_stream = nullptr; /* rate may have changed, any music source is now invalid */
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return true;
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}
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void SetEffectVolume(uint8 volume)
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{
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_effect_vol.store(volume, std::memory_order_relaxed);
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}
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