mirror of
https://github.com/kanjitalk755/macemu.git
synced 2024-10-14 02:24:02 +00:00
added format switching and other improvements [Brian Johnson]
This commit is contained in:
cebix
parent
0074b3346e
commit
16b21014ef
@ -40,6 +40,12 @@
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#include "debug.h"
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// The currently selected audio parameters (indices in audio_sample_rates[]
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// etc. vectors)
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static int audio_sample_rate_index = 0;
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static int audio_sample_size_index = 0;
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static int audio_channel_count_index = 0;
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// Global variables
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static int audio_fd = -1; // fd from audio library
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static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
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@ -52,6 +58,11 @@ static pthread_attr_t stream_thread_attr; // Streaming thread attributes
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static bool stream_thread_active = false; // Flag: streaming thread installed
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static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
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static bool current_main_mute = false; // Flag: output muted
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static bool current_speaker_mute = false; // Flag: speaker muted
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static uint32 current_main_volume = 0; // Output volume
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static uint32 current_speaker_volume = 0; // Speaker volume
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// IRIX libaudio control structures
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static ALconfig config;
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static ALport port;
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@ -59,6 +70,9 @@ static ALport port;
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// Prototypes
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static void *stream_func(void *arg);
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static uint32 read_volume(void);
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static bool read_mute(void);
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static void set_mute(bool mute);
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/*
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@ -68,37 +82,78 @@ static void *stream_func(void *arg);
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// Set AudioStatus to reflect current audio stream format
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static void set_audio_status_format(void)
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{
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AudioStatus.sample_rate = audio_sample_rates[0];
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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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AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
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AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
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AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
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}
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// Init libaudio, returns false on error
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bool audio_init_al(void)
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bool open_audio(void)
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{
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ALpv pv[2];
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printf("Using libaudio audio output\n");
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// Try to open the audio library
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// Get supported sample formats
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if (audio_sample_sizes.empty()) {
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// All sample sizes are supported
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audio_sample_sizes.push_back(8);
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audio_sample_sizes.push_back(16);
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// Assume at least two channels are supported. Some IRIX boxes
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// can do 4 or more... MacOS only handles up to 2.
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audio_channel_counts.push_back(1);
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audio_channel_counts.push_back(2);
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if (audio_sample_sizes.empty() || audio_channel_counts.empty()) {
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WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
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alClosePort(port);
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audio_fd = -1;
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return false;
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}
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audio_sample_rates.push_back( 8000 << 16);
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audio_sample_rates.push_back(11025 << 16);
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audio_sample_rates.push_back(22050 << 16);
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audio_sample_rates.push_back(44100 << 16);
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// Default to highest supported values
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audio_sample_rate_index = audio_sample_rates.size() - 1;
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audio_sample_size_index = audio_sample_sizes.size() - 1;
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audio_channel_count_index = audio_channel_counts.size() - 1;
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}
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// Set the sample format
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D(bug("Size %d, channels %d, rate %d\n",
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audio_sample_sizes[audio_sample_size_index],
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audio_channel_counts[audio_channel_count_index],
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audio_sample_rates[audio_sample_rate_index] >> 16));
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config = alNewConfig();
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alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
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alSetWidth(config, AL_SAMPLE_16);
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alSetChannels(config, 2); // stereo
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if (audio_sample_sizes[audio_sample_size_index] == 8) {
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alSetWidth(config, AL_SAMPLE_8);
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}
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else {
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alSetWidth(config, AL_SAMPLE_16);
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}
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alSetChannels(config, audio_channel_counts[audio_channel_count_index]);
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alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
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// Try to open the audio library
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port = alOpenPort("BasiliskII", "w", config);
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if (port == NULL) {
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fprintf(stderr, "ERROR: Cannot open audio port: %s\n",
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alGetErrorString(oserror()));
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WarningAlert(GetString(STR_NO_AUDIO_WARN));
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return false;
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}
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// Set the sample rate
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pv[0].param = AL_RATE;
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pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
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pv[0].value.ll = alDoubleToFixed(audio_sample_rates[audio_sample_rate_index] >> 16);
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pv[1].param = AL_MASTER_CLOCK;
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pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
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if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
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@ -108,12 +163,6 @@ bool audio_init_al(void)
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return false;
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}
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// TODO: list all supported sample formats?
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// Set AudioStatus again because we now know more about the sound
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// system's capabilities
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set_audio_status_format();
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// Compute sound buffer size and libaudio refill point
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config = alGetConfig(port);
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@ -124,20 +173,33 @@ bool audio_init_al(void)
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alClosePort(port);
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return false;
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}
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D(bug("alGetQueueSize %d\n", audio_frames_per_block));
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D(bug("alGetQueueSize %d, width %d, channels %d\n",
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audio_frames_per_block,
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alGetWidth(config),
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alGetChannels(config)));
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// Put a limit on the Mac sound buffer size, to decrease delay
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#define AUDIO_BUFFER_MSEC 50 // milliseconds of sound to buffer
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int target_frames_per_block =
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(audio_sample_rates[audio_sample_rate_index] >> 16) *
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AUDIO_BUFFER_MSEC / 1000;
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if (audio_frames_per_block > target_frames_per_block)
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audio_frames_per_block = target_frames_per_block;
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D(bug("frames per block %d\n", audio_frames_per_block));
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alZeroFrames(port, audio_frames_per_block); // so we don't underflow
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// Put a limit on the Mac sound buffer size, to decrease delay
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if (audio_frames_per_block > 2048)
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audio_frames_per_block = 2048;
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// Try to keep the buffer pretty full. 5000 samples of slack works well.
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sound_buffer_fill_point = alGetQueueSize(config) - 5000;
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// Try to keep the buffer pretty full
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sound_buffer_fill_point = alGetQueueSize(config) -
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2 * audio_frames_per_block;
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if (sound_buffer_fill_point < 0)
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sound_buffer_fill_point = alGetQueueSize(config) / 3;
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D(bug("fill point %d\n", sound_buffer_fill_point));
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sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
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sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) *
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audio_channel_counts[audio_channel_count_index] *
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audio_frames_per_block;
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set_audio_status_format();
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// Get a file descriptor we can select() on
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@ -149,21 +211,26 @@ bool audio_init_al(void)
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return false;
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}
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// Initialize volume, mute settings
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current_main_volume = current_speaker_volume = read_volume();
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current_main_mute = current_speaker_mute = read_mute();
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// Start streaming thread
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Set_pthread_attr(&stream_thread_attr, 0);
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stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
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// Everything went fine
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audio_open = true;
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return true;
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}
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/*
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* Initialization
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*/
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void AudioInit(void)
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{
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// Init audio status (defaults) and feature flags
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audio_sample_rates.push_back(44100 << 16);
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audio_sample_sizes.push_back(16);
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audio_channel_counts.push_back(2);
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set_audio_status_format();
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// Init audio status (reasonable defaults) and feature flags
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AudioStatus.sample_rate = 44100 << 16;
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AudioStatus.sample_size = 16;
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AudioStatus.channels = 2;
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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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@ -172,21 +239,13 @@ void AudioInit(void)
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if (PrefsFindBool("nosound"))
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return;
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// Try to open audio library
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if (!audio_init_al())
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return;
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// Init semaphore
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if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
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return;
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sem_inited = true;
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// Start streaming thread
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Set_pthread_attr(&stream_thread_attr, 0);
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stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
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// Everything OK
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audio_open = true;
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// Open and initialize audio device
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open_audio();
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}
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@ -194,7 +253,7 @@ void AudioInit(void)
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* Deinitialization
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*/
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void AudioExit(void)
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static void close_audio(void)
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{
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// Stop stream and delete semaphore
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if (stream_thread_active) {
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@ -204,12 +263,25 @@ void AudioExit(void)
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#endif
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pthread_join(stream_thread, NULL);
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stream_thread_active = false;
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stream_thread_cancel = false;
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}
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if (sem_inited)
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sem_destroy(&audio_irq_done_sem);
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// Close audio library
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alClosePort(port);
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audio_open = false;
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}
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void AudioExit(void)
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{
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// Close audio device
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close_audio();
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// Delete semaphore
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if (sem_inited) {
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sem_destroy(&audio_irq_done_sem);
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sem_inited = false;
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}
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}
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@ -239,7 +311,7 @@ void audio_exit_stream()
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static void *stream_func(void *arg)
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{
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int16 *last_buffer = new int16[sound_buffer_size / 2];
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int32 *last_buffer = new int32[sound_buffer_size / 4];
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fd_set audio_fdset;
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int numfds, was_error;
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@ -257,11 +329,11 @@ static void *stream_func(void *arg)
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sem_wait(&audio_irq_done_sem);
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D(bug("stream: ack received\n"));
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uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
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// Get size of audio data
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apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
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if (apple_stream_info) {
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uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
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if (!current_main_mute &&
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!current_speaker_mute &&
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apple_stream_info) {
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int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
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D(bug("stream: work_size %d\n", work_size));
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if (work_size > sound_buffer_size)
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@ -269,8 +341,21 @@ static void *stream_func(void *arg)
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if (work_size == 0)
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goto silence;
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// Send data to audio library
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if (work_size == sound_buffer_size)
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// Send data to audio library. Convert 8-bit data
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// unsigned->signed, using same algorithm as audio_amiga.cpp.
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// It works fine for 8-bit mono, but not stereo.
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if (AudioStatus.sample_size == 8) {
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uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
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uint32 *q = (uint32 *)last_buffer;
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int r = work_size >> 2;
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// XXX not quite right....
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while (r--)
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*q++ = *p++ ^ 0x80808080;
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if (work_size != sound_buffer_size)
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memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
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alWriteFrames(port, last_buffer, audio_frames_per_block);
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}
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else if (work_size == sound_buffer_size)
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alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
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else {
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// Last buffer
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@ -313,6 +398,156 @@ static void *stream_func(void *arg)
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}
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/*
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* Read or set the current output volume using the audio library
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*/
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static uint32 read_volume(void)
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{
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ALpv x[2];
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ALfixed gain[8];
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double maxgain, mingain;
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ALparamInfo pi;
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uint32 ret = 0x01000100; // default, maximum value
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int dev = alGetDevice(config);
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// Fetch the maximum and minimum gain settings
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alGetParamInfo(dev, AL_GAIN, &pi);
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maxgain = alFixedToDouble(pi.max.ll);
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mingain = alFixedToDouble(pi.min.ll);
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// printf("maxgain = %lf dB, mingain = %lf dB\n", maxgain, mingain);
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// Get the current gain values
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x[0].param = AL_GAIN;
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x[0].value.ptr = gain;
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x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);
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x[1].param = AL_CHANNELS;
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if (alGetParams(dev, x, 2) < 0) {
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printf("alGetParams failed: %s\n", alGetErrorString(oserror()));
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}
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else {
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if (x[0].sizeOut < 0) {
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printf("AL_GAIN was an unrecognized parameter\n");
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}
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else {
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double v;
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uint32 left, right;
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// Left
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v = alFixedToDouble(gain[0]);
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if (v < mingain)
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v = mingain; // handle gain == -inf
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v = (v - mingain) / (maxgain - mingain); // scale to 0..1
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left = (uint32)(v * (double)256); // convert to 8.8 fixed point
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// Right
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if (x[0].sizeOut <= 1) { // handle a mono interface
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right = left;
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}
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else {
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v = alFixedToDouble(gain[1]);
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if (v < mingain)
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v = mingain; // handle gain == -inf
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v = (v - mingain) / (maxgain - mingain); // scale to 0..1
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right = (uint32)(v * (double)256); // convert to 8.8 fixed point
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}
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ret = (left << 16) | right;
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}
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}
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return ret;
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}
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static void set_volume(uint32 vol)
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{
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ALpv x[1];
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ALfixed gain[2]; // left and right
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double maxgain, mingain;
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ALparamInfo pi;
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int dev = alGetDevice(config);
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// Fetch the maximum and minimum gain settings
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alGetParamInfo(dev, AL_GAIN, &pi);
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maxgain = alFixedToDouble(pi.max.ll);
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mingain = alFixedToDouble(pi.min.ll);
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// Set the new gain values
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x[0].param = AL_GAIN;
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x[0].value.ptr = gain;
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x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);
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uint32 left = vol >> 16;
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uint32 right = vol & 0xffff;
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double lv, rv;
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if (left == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
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lv = AL_NEG_INFINITY;
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}
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else {
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lv = ((double)left / 256) * (maxgain - mingain) + mingain;
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}
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if (right == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
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rv = AL_NEG_INFINITY;
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}
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else {
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rv = ((double)right / 256) * (maxgain - mingain) + mingain;
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}
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D(bug("set_volume: left=%lf dB, right=%lf dB\n", lv, rv));
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gain[0] = alDoubleToFixed(lv);
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gain[1] = alDoubleToFixed(rv);
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if (alSetParams(dev, x, 1) < 0) {
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printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
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}
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}
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/*
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* Read or set the mute setting using the audio library
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*/
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static bool read_mute(void)
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{
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bool ret;
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int dev = alGetDevice(config);
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ALpv x;
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x.param = AL_MUTE;
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if (alGetParams(dev, &x, 1) < 0) {
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printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
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return current_main_mute; // Or just return false?
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}
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ret = x.value.i;
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D(bug("read_mute: mute=%d\n", ret));
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return ret;
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}
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static void set_mute(bool mute)
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{
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D(bug("set_mute: mute=%ld\n", mute));
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int dev = alGetDevice(config);
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ALpv x;
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x.param = AL_MUTE;
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x.value.i = mute;
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||||
|
||||
if (alSetParams(dev, &x, 1) < 0) {
|
||||
printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
|
||||
/*
|
||||
* MacOS audio interrupt, read next data block
|
||||
*/
|
||||
@ -339,23 +574,29 @@ void AudioInterrupt(void)
|
||||
|
||||
/*
|
||||
* Set sampling parameters
|
||||
* "index" is an index into the audio_sample_rates[] etc. arrays
|
||||
* "index" is an index into the audio_sample_rates[] etc. vectors
|
||||
* It is guaranteed that AudioStatus.num_sources == 0
|
||||
*/
|
||||
|
||||
bool audio_set_sample_rate(int index)
|
||||
{
|
||||
return true;
|
||||
close_audio();
|
||||
audio_sample_rate_index = index;
|
||||
return open_audio();
|
||||
}
|
||||
|
||||
bool audio_set_sample_size(int index)
|
||||
{
|
||||
return true;
|
||||
close_audio();
|
||||
audio_sample_size_index = index;
|
||||
return open_audio();
|
||||
}
|
||||
|
||||
bool audio_set_channels(int index)
|
||||
{
|
||||
return true;
|
||||
close_audio();
|
||||
audio_channel_count_index = index;
|
||||
return open_audio();
|
||||
}
|
||||
|
||||
|
||||
@ -367,36 +608,73 @@ bool audio_set_channels(int index)
|
||||
|
||||
bool audio_get_main_mute(void)
|
||||
{
|
||||
return false;
|
||||
D(bug("audio_get_main_mute: mute=%ld\n", current_main_mute));
|
||||
|
||||
return current_main_mute;
|
||||
}
|
||||
|
||||
uint32 audio_get_main_volume(void)
|
||||
{
|
||||
return 0x01000100;
|
||||
uint32 ret = current_main_volume;
|
||||
|
||||
D(bug("audio_get_main_volume: vol=0x%x\n", ret));
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
bool audio_get_speaker_mute(void)
|
||||
{
|
||||
return false;
|
||||
D(bug("audio_get_speaker_mute: mute=%ld\n", current_speaker_mute));
|
||||
|
||||
return current_speaker_mute;
|
||||
}
|
||||
|
||||
uint32 audio_get_speaker_volume(void)
|
||||
{
|
||||
return 0x01000100;
|
||||
uint32 ret = current_speaker_volume;
|
||||
|
||||
D(bug("audio_get_speaker_volume: vol=0x%x\n", ret));
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
void audio_set_main_mute(bool mute)
|
||||
{
|
||||
D(bug("audio_set_main_mute: mute=%ld\n", mute));
|
||||
|
||||
if (mute != current_main_mute) {
|
||||
current_main_mute = mute;
|
||||
}
|
||||
|
||||
set_mute(current_main_mute);
|
||||
}
|
||||
|
||||
void audio_set_main_volume(uint32 vol)
|
||||
{
|
||||
|
||||
D(bug("audio_set_main_volume: vol=%x\n", vol));
|
||||
|
||||
current_main_volume = vol;
|
||||
|
||||
set_volume(vol);
|
||||
}
|
||||
|
||||
void audio_set_speaker_mute(bool mute)
|
||||
{
|
||||
D(bug("audio_set_speaker_mute: mute=%ld\n", mute));
|
||||
|
||||
if (mute != current_speaker_mute) {
|
||||
current_speaker_mute = mute;
|
||||
}
|
||||
|
||||
set_mute(current_speaker_mute);
|
||||
}
|
||||
|
||||
void audio_set_speaker_volume(uint32 vol)
|
||||
{
|
||||
D(bug("audio_set_speaker_volume: vol=%x\n", vol));
|
||||
|
||||
current_speaker_volume = vol;
|
||||
|
||||
set_volume(vol);
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user