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root/cebix/BasiliskII/src/Unix/audio_oss_esd.cpp
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Comparing BasiliskII/src/Unix/audio_oss_esd.cpp (file contents):
Revision 1.1 by cebix, 1999-10-23T17:57:45Z vs.
Revision 1.2 by cebix, 1999-10-25T19:01:34Z

# Line 49 | Line 49
49   #include "debug.h"
50  
51  
52 < // Supported sample rates, sizes and channels
52 > // Supported sample rates, sizes and channels (defaults)
53   int audio_num_sample_rates = 1;
54   uint32 audio_sample_rates[] = {44100 << 16};
55   int audio_num_sample_sizes = 1;
# Line 67 | Line 67 | static sem_t audio_irq_done_sem;                                       //
67   static bool sem_inited = false;                                         // Flag: audio_irq_done_sem initialized
68   static int sound_buffer_size;                                           // Size of sound buffer in bytes
69   static bool little_endian = false;                                      // Flag: DSP accepts only little-endian 16-bit sound data
70 + static uint8 silence_byte;                                                      // Byte value to use to fill sound buffers with silence
71   static pthread_t stream_thread;                                         // Audio streaming thread
72   static pthread_attr_t stream_thread_attr;                       // Streaming thread attributes
73   static bool stream_thread_active = false;                       // Flag: streaming thread installed
# Line 80 | Line 81 | static void *stream_func(void *arg);
81   *  Initialization
82   */
83  
84 + // Set AudioStatus to reflect current audio stream format
85 + static void set_audio_status_format(void)
86 + {
87 +        AudioStatus.sample_rate = audio_sample_rates[0];
88 +        AudioStatus.sample_size = audio_sample_sizes[0];
89 +        AudioStatus.channels = audio_channel_counts[0];
90 + }
91 +
92   // Init using /dev/dsp, returns false on error
93   bool audio_init_dsp(void)
94   {
# Line 94 | Line 103 | bool audio_init_dsp(void)
103                  audio_fd = -1;
104                  return false;
105          }
106 <        if (format & (AFMT_S16_BE | AFMT_S16_LE))
106 >        if (format & (AFMT_S16_BE | AFMT_S16_LE)) {
107                  audio_sample_sizes[0] = 16;
108 <        else
108 >                silence_byte = 0;
109 >        } else {
110                  audio_sample_sizes[0] = 8;
111 +                silence_byte = 0x80;
112 +        }
113          if (!(format & AFMT_S16_BE))
114                  little_endian = true;
115  
116          // Set DSP parameters
117 <        format = AudioStatus.sample_size == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
117 >        format = audio_sample_sizes[0] == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
118          ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
119          int frag = 0x0004000c;          // Block size: 4096 frames
120          ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
121 <        int stereo = (AudioStatus.channels == 2);
121 >        int stereo = (audio_channel_counts[0] == 2);
122          ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
123 <        int rate = AudioStatus.sample_rate >> 16;
123 >        int rate = audio_sample_rates[0] >> 16;
124          ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
125 +        audio_sample_rates[0] = rate << 16;
126 +
127 +        // Set AudioStatus again because we now know more about the sound
128 +        // system's capabilities
129 +        set_audio_status_format();
130  
131          // Get sound buffer size
132          ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
# Line 140 | Line 157 | bool audio_init_esd(void)
157   #else
158          little_endian = true;
159   #endif
160 +        silence_byte = 0;       // Is this correct for 8-bit mode?
161  
162          // Open connection to ESD server
163          audio_fd = esd_play_stream(format, AudioStatus.sample_rate >> 16, NULL, NULL);
# Line 165 | Line 183 | void AudioInit(void)
183          char str[256];
184  
185          // Init audio status (defaults) and feature flags
186 <        AudioStatus.sample_rate = audio_sample_rates[0];
169 <        AudioStatus.sample_size = audio_sample_sizes[0];
170 <        AudioStatus.channels = audio_channel_counts[0];
186 >        set_audio_status_format();
187          AudioStatus.mixer = 0;
188          AudioStatus.num_sources = 0;
189          audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;
# Line 277 | Line 293 | static void *stream_func(void *arg)
293   {
294          int16 *silent_buffer = new int16[sound_buffer_size / 2];
295          int16 *last_buffer = new int16[sound_buffer_size / 2];
296 <        memset(silent_buffer, 0, sound_buffer_size);
296 >        memset(silent_buffer, silence_byte, sound_buffer_size);
297  
298          while (!stream_thread_cancel) {
299                  if (AudioStatus.num_sources) {
# Line 311 | Line 327 | static void *stream_func(void *arg)
327                                                          last_buffer[i] = ntohs(p[i]);
328                                          } else
329                                                  memcpy(last_buffer, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
330 <                                        memset((uint8 *)last_buffer + work_size, 0, sound_buffer_size - work_size);
330 >                                        memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
331                                          write(audio_fd, last_buffer, sound_buffer_size);
332                                  }
333                                  D(bug("stream: data written\n"));

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