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/* |
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* audio_amiga.cpp - Audio support, AmigaOS implementation using AHI |
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* |
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* Basilisk II (C) 1997-2001 Christian Bauer |
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* |
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* This program is free software; you can redistribute it and/or modify |
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* it under the terms of the GNU General Public License as published by |
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* the Free Software Foundation; either version 2 of the License, or |
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* (at your option) any later version. |
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* |
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* This program is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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* GNU General Public License for more details. |
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* |
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* You should have received a copy of the GNU General Public License |
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* along with this program; if not, write to the Free Software |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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*/ |
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|
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#include "sysdeps.h" |
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|
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#include <exec/types.h> |
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#include <exec/memory.h> |
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#include <devices/ahi.h> |
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#define __USE_SYSBASE |
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#include <proto/exec.h> |
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#include <proto/ahi.h> |
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#include <inline/exec.h> |
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#include <inline/ahi.h> |
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|
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#include "cpu_emulation.h" |
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#include "main.h" |
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#include "prefs.h" |
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#include "user_strings.h" |
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#include "audio.h" |
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#include "audio_defs.h" |
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|
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#define DEBUG 0 |
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#include "debug.h" |
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|
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#define D1(x) ; |
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|
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|
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// Global variables |
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static ULONG ahi_id = AHI_DEFAULT_ID; // AHI audio ID |
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static struct AHIAudioCtrl *ahi_ctrl = NULL; |
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static struct AHISampleInfo sample[2]; // Two sample infos for double-buffering |
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static struct Hook sf_hook; |
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static int play_buf = 0; // Number of currently played buffer |
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static long sound_buffer_size; // Size of one audio buffer in bytes |
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static int audio_block_fetched = 0; // Number of audio blocks fetched by interrupt routine |
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|
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static bool main_mute = false; |
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static bool speaker_mute = false; |
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static ULONG supports_volume_changes = false; |
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static ULONG supports_stereo_panning = false; |
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static ULONG current_main_volume; |
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static ULONG current_speaker_volume; |
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|
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|
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// Prototypes |
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static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/); |
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void audio_set_sample_rate_byval(uint32 value); |
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void audio_set_sample_size_byval(uint32 value); |
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void audio_set_channels_byval(uint32 value); |
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|
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|
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/* |
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* Initialization |
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*/ |
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|
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// Set AudioStatus to reflect current audio stream format |
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static void set_audio_status_format(int sample_rate_index) |
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{ |
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AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; |
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AudioStatus.sample_size = audio_sample_sizes[0]; |
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AudioStatus.channels = audio_channel_counts[0]; |
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} |
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|
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void AudioInit(void) |
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{ |
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sample[0].ahisi_Address = sample[1].ahisi_Address = NULL; |
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|
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// Init audio status and feature flags |
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audio_channel_counts.push_back(2); |
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// set_audio_status_format(); |
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AudioStatus.mixer = 0; |
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AudioStatus.num_sources = 0; |
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audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut; |
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|
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// Sound disabled in prefs? Then do nothing |
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if (PrefsFindBool("nosound")) |
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return; |
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|
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// AHI available? |
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if (AHIBase == NULL) { |
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WarningAlert(GetString(STR_NO_AHI_WARN)); |
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return; |
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} |
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|
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// Initialize callback hook |
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sf_hook.h_Entry = (HOOKFUNC)audio_callback; |
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|
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// Read "sound" preferences |
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const char *str = PrefsFindString("sound"); |
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if (str) |
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sscanf(str, "ahi/%08lx", &ahi_id); |
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|
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// Open audio control structure |
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if ((ahi_ctrl = AHI_AllocAudio( |
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AHIA_AudioID, ahi_id, |
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AHIA_MixFreq, AudioStatus.sample_rate >> 16, |
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AHIA_Channels, 1, |
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AHIA_Sounds, 2, |
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AHIA_SoundFunc, (ULONG)&sf_hook, |
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TAG_END)) == NULL) { |
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WarningAlert(GetString(STR_NO_AHI_CTRL_WARN)); |
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return; |
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} |
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|
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ULONG max_channels, sample_rate, frequencies, sample_rate_index; |
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|
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_MaxChannels, (ULONG) &max_channels, |
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AHIDB_Frequencies, (ULONG) &frequencies, |
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TAG_END); |
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|
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D(bug("AudioInit: max_channels=%ld frequencies=%ld\n", max_channels, frequencies)); |
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|
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for (int n=0; n<frequencies; n++) |
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{ |
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_FrequencyArg, n, |
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AHIDB_Frequency, (ULONG) &sample_rate, |
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TAG_END); |
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|
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D(bug("AudioInit: f=%ld Hz\n", sample_rate)); |
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audio_sample_rates.push_back(sample_rate << 16); |
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} |
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|
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ULONG sample_size_bits = 16; |
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|
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D(bug("AudioInit: sampe_rates=%ld\n", audio_sample_rates.size() )); |
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|
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// get index of sample rate closest to 22050 Hz |
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_IndexArg, 22050, |
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AHIDB_Bits, (ULONG) &sample_size_bits, |
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AHIDB_Index, (ULONG) &sample_rate_index, |
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AHIDB_Volume, (ULONG) &supports_volume_changes, |
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AHIDB_Panning, (ULONG) &supports_stereo_panning, |
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TAG_END); |
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|
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audio_sample_sizes.push_back(16); |
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|
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set_audio_status_format(sample_rate_index); |
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|
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// 2048 frames per block |
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audio_frames_per_block = 2048; |
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sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block; |
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|
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// Prepare SampleInfos and load sounds (two sounds for double buffering) |
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sample[0].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S; |
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sample[0].ahisi_Length = audio_frames_per_block; |
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sample[0].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR); |
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sample[1].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S; |
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sample[1].ahisi_Length = audio_frames_per_block; |
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sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR); |
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if (sample[0].ahisi_Address == NULL || sample[1].ahisi_Address == NULL) |
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return; |
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|
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AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl); |
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AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl); |
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|
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// Set parameters |
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play_buf = 0; |
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current_main_volume = current_speaker_volume = 0x10000; |
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AHI_SetVol(0, current_speaker_volume, 0x8000, ahi_ctrl, AHISF_IMM); |
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|
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AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM); |
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AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM); |
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|
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// Everything OK |
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audio_open = true; |
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} |
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|
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|
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/* |
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* Deinitialization |
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*/ |
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|
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void AudioExit(void) |
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{ |
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// Free everything |
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if (ahi_ctrl != NULL) { |
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AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END); |
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AHI_FreeAudio(ahi_ctrl); |
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} |
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|
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FreeVec(sample[0].ahisi_Address); |
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FreeVec(sample[1].ahisi_Address); |
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} |
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|
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|
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/* |
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* First source added, start audio stream |
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*/ |
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|
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void audio_enter_stream() |
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{ |
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AHI_ControlAudio(ahi_ctrl, AHIC_Play, TRUE, TAG_END); |
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} |
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|
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|
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/* |
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* Last source removed, stop audio stream |
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*/ |
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|
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void audio_exit_stream() |
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{ |
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AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END); |
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} |
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|
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|
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/* |
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* AHI sound callback, request next buffer |
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*/ |
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|
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static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/) |
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{ |
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play_buf ^= 1; |
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|
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// New buffer available? |
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if (audio_block_fetched) |
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{ |
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audio_block_fetched--; |
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|
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if (main_mute || speaker_mute) |
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{ |
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memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); |
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} |
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else |
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{ |
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// Get size of audio data |
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uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo); |
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if (apple_stream_info) { |
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int32 sample_count = ReadMacInt32(apple_stream_info + scd_sampleCount); |
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|
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uint32 num_channels = ReadMacInt16(apple_stream_info + scd_numChannels); |
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uint32 sample_size = ReadMacInt16(apple_stream_info + scd_sampleSize); |
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uint32 sample_rate = ReadMacInt32(apple_stream_info + scd_sampleRate); |
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|
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D(bug("stream: sample_count=%ld num_channels=%ld sample_size=%ld sample_rate=%ld\n", sample_count, num_channels, sample_size, sample_rate >> 16)); |
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|
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// Yes, this can happen. |
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if(sample_count != 0) { |
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if(sample_rate != AudioStatus.sample_rate) { |
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audio_set_sample_rate_byval(sample_rate); |
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} |
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if(num_channels != AudioStatus.channels) { |
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audio_set_channels_byval(num_channels); |
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} |
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if(sample_size != AudioStatus.sample_size) { |
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audio_set_sample_size_byval(sample_size); |
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} |
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} |
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|
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if (sample_count < 0) |
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sample_count = 0; |
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|
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int work_size = sample_count * num_channels * (sample_size>>3); |
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D(bug("stream: work_size=%ld sound_buffer_size=%ld\n", work_size, sound_buffer_size)); |
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|
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if (work_size > sound_buffer_size) |
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work_size = sound_buffer_size; |
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|
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// Put data into AHI buffer (convert 8-bit data unsigned->signed) |
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if (AudioStatus.sample_size == 16) |
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Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size); |
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else { |
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uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)); |
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uint32 *q = (uint32 *)sample[play_buf].ahisi_Address; |
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int r = work_size >> 2; |
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while (r--) |
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*q++ = *p++ ^ 0x80808080; |
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} |
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if (work_size != sound_buffer_size) |
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memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size); |
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} |
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} |
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|
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} |
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else |
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memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); |
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|
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// Play next buffer |
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AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, 0); |
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|
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// Trigger audio interrupt to get new buffer |
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if (AudioStatus.num_sources) { |
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D1(bug("stream: triggering irq\n")); |
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SetInterruptFlag(INTFLAG_AUDIO); |
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TriggerInterrupt(); |
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} |
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return 0; |
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} |
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|
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|
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/* |
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* MacOS audio interrupt, read next data block |
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*/ |
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|
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void AudioInterrupt(void) |
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{ |
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D1(bug("AudioInterrupt\n")); |
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|
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// Get data from apple mixer |
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if (AudioStatus.mixer) { |
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M68kRegisters r; |
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r.a[0] = audio_data + adatStreamInfo; |
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r.a[1] = AudioStatus.mixer; |
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Execute68k(audio_data + adatGetSourceData, &r); |
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D1(bug(" GetSourceData() returns %08lx\n", r.d[0])); |
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} else |
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WriteMacInt32(audio_data + adatStreamInfo, 0); |
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|
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// Signal stream function |
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audio_block_fetched++; |
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D1(bug("AudioInterrupt done\n")); |
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} |
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|
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|
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/* |
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* Set sampling parameters |
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* "index" is an index into the audio_sample_rates[] etc. arrays |
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* It is guaranteed that AudioStatus.num_sources == 0 |
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*/ |
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|
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void audio_set_sample_rate_byval(uint32 value) |
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{ |
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bool changed = (AudioStatus.sample_rate != value); |
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if(changed) |
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{ |
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ULONG sample_rate_index; |
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|
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// get index of sample rate closest to <value> Hz |
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_IndexArg, value >> 16, |
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AHIDB_Index, (ULONG) &sample_rate_index, |
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TAG_END); |
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|
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D(bug(" audio_set_sample_rate_byval requested rate=%ld Hz\n", value >> 16)); |
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|
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AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; |
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|
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AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, 0); |
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} |
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|
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D(bug(" audio_set_sample_rate_byval rate=%ld Hz\n", AudioStatus.sample_rate >> 16)); |
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} |
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|
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void audio_set_sample_size_byval(uint32 value) |
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{ |
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bool changed = (AudioStatus.sample_size != value); |
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if(changed) { |
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// AudioStatus.sample_size = value; |
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// update_sound_parameters(); |
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// WritePrivateProfileInt( "Audio", "SampleSize", AudioStatus.sample_size, ini_file_name ); |
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} |
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D(bug(" audio_set_sample_size_byval %d\n", AudioStatus.sample_size)); |
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} |
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|
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void audio_set_channels_byval(uint32 value) |
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{ |
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bool changed = (AudioStatus.channels != value); |
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if(changed) { |
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// AudioStatus.channels = value; |
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// update_sound_parameters(); |
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// WritePrivateProfileInt( "Audio", "Channels", AudioStatus.channels, ini_file_name ); |
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} |
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D(bug(" audio_set_channels_byval %d\n", AudioStatus.channels)); |
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} |
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|
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bool audio_set_sample_rate(int index) |
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{ |
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if(index >= 0 && index < audio_sample_rates.size() ) { |
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audio_set_sample_rate_byval( audio_sample_rates[index] ); |
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D(bug(" audio_set_sample_rate index=%ld rate=%ld\n", index, AudioStatus.sample_rate >> 16)); |
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} |
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|
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return true; |
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} |
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|
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bool audio_set_sample_size(int index) |
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{ |
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if(index >= 0 && index < audio_sample_sizes.size() ) { |
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audio_set_sample_size_byval( audio_sample_sizes[index] ); |
399 |
D(bug(" audio_set_sample_size %d,%d\n", index,AudioStatus.sample_size)); |
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} |
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|
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return true; |
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} |
404 |
|
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bool audio_set_channels(int index) |
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{ |
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if(index >= 0 && index < audio_channel_counts.size() ) { |
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audio_set_channels_byval( audio_channel_counts[index] ); |
409 |
D(bug(" audio_set_channels %d,%d\n", index,AudioStatus.channels)); |
410 |
} |
411 |
|
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return true; |
413 |
} |
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|
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|
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/* |
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* Get/set volume controls (volume values received/returned have the left channel |
418 |
* volume in the upper 16 bits and the right channel volume in the lower 16 bits; |
419 |
* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume")) |
420 |
*/ |
421 |
|
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bool audio_get_main_mute(void) |
423 |
{ |
424 |
D(bug("audio_get_main_mute: mute=%ld\n", main_mute)); |
425 |
|
426 |
return main_mute; |
427 |
} |
428 |
|
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uint32 audio_get_main_volume(void) |
430 |
{ |
431 |
D(bug("audio_get_main_volume\n")); |
432 |
|
433 |
ULONG volume = current_main_volume >> 8; // 0x10000 => 0x100 |
434 |
|
435 |
D(bug("audio_get_main_volume: volume=%08lx\n", volume)); |
436 |
|
437 |
return (volume << 16) + volume; |
438 |
|
439 |
return 0x01000100; |
440 |
} |
441 |
|
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bool audio_get_speaker_mute(void) |
443 |
{ |
444 |
D(bug("audio_get_speaker_mute: mute=%ld\n", speaker_mute)); |
445 |
|
446 |
return speaker_mute; |
447 |
} |
448 |
|
449 |
uint32 audio_get_speaker_volume(void) |
450 |
{ |
451 |
D(bug("audio_get_speaker_volume: \n")); |
452 |
|
453 |
if (audio_open) |
454 |
{ |
455 |
ULONG volume = current_speaker_volume >> 8; // 0x10000 => 0x100 |
456 |
|
457 |
D(bug("audio_get_speaker_volume: volume=%08lx\n", volume)); |
458 |
|
459 |
return (volume << 16) + volume; |
460 |
} |
461 |
|
462 |
return 0x01000100; |
463 |
} |
464 |
|
465 |
void audio_set_main_mute(bool mute) |
466 |
{ |
467 |
D(bug("audio_set_main_mute: mute=%ld\n", mute)); |
468 |
|
469 |
if (mute != main_mute) |
470 |
{ |
471 |
main_mute = mute; |
472 |
} |
473 |
} |
474 |
|
475 |
void audio_set_main_volume(uint32 vol) |
476 |
{ |
477 |
D(bug("audio_set_main_volume: vol=%08lx\n", vol)); |
478 |
|
479 |
if (audio_open && supports_volume_changes) |
480 |
{ |
481 |
ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); |
482 |
|
483 |
D(bug("audio_set_main_volume: volume=%08lx\n", volume)); |
484 |
|
485 |
current_main_volume = volume; |
486 |
|
487 |
AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); |
488 |
} |
489 |
} |
490 |
|
491 |
void audio_set_speaker_mute(bool mute) |
492 |
{ |
493 |
D(bug("audio_set_speaker_mute: mute=%ld\n", mute)); |
494 |
|
495 |
if (mute != speaker_mute) |
496 |
{ |
497 |
speaker_mute = mute; |
498 |
} |
499 |
} |
500 |
|
501 |
void audio_set_speaker_volume(uint32 vol) |
502 |
{ |
503 |
D(bug("audio_set_speaker_volume: vol=%08lx\n", vol)); |
504 |
|
505 |
if (audio_open && supports_volume_changes) |
506 |
{ |
507 |
ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); |
508 |
|
509 |
D(bug("audio_set_speaker_volume: volume=%08lx\n", volume)); |
510 |
|
511 |
current_speaker_volume = volume; |
512 |
|
513 |
AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); |
514 |
} |
515 |
} |