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* ddksample_audio.c - OSS DDK sample driver - Audio functionality
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* This simple audio driver implements an OSS audio device similar to
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* /dev/zero. The output will be just discarded. Recording will return
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* silence. In addition the sample parameters (rate, channels and format)
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* is emulated. Timeout callbacks are used to keep the device running
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* at the right rate (approximately).
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* To demonstrate mixer functionality (together with ddksample_mixer.c) this
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* driver also computes the peak levels from the output data. Also output
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* volume controls are implemented.
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* This file is part of Open Sound System.
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* Copyright (C) 4Front Technologies 1996-2008.
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* This this source file is released under GPL v2 license (no other versions).
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* See the COPYING file included in the main directory of this source
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* distribution for the license terms and conditions.
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* Solaris DDI includes
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#include <sys/types.h>
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#include <sys/modctl.h>
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#include <sys/sunddi.h>
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* OSS specific includes
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#include <sys/soundcard.h>
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#include <sys/ossddk/ossddk.h>
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/***************************************************
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* Private types and variables
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#include "ddksample.h"
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/**************************************************/
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* This driver emulates periodic interrupts on fragment boundaries using
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* the timeout(9f) mechanism. Ordinary drivers use device interrupts
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ddksample_intr (void *arg)
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ddksample_portc *portc = arg;
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if (portc->audio_active & PCM_ENABLE_OUTPUT)
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* We will first do some computations on the output data. This is not
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* necessary in normal drivers. It's here just to make the mixer section
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* to behave like the mixer of some real device.
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dmap = ossddk_adev_get_dmapout (portc->dev);
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buf = ossddk_dmap_get_dmabuf (dmap);
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* Compute the current playback position (beginning of the
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* fragment we have just finished).
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pos = ossddk_dmap_get_qhead (dmap) * ossddk_dmap_get_fragsize (dmap);
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ddksample_do_math (portc, buf, ossddk_dmap_get_fragsize (dmap));
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ossddk_audio_outputintr (portc->dev, 0); /* Acknowledge one fragment */
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if (portc->audio_active & PCM_ENABLE_INPUT)
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ossddk_audio_inputintr (portc->dev); /* Acknowledge one fragment */
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portc->timeout_id = timeout (ddksample_intr, portc, portc->timeout_value);
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ddksample_set_rate (int dev, int arg)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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if (arg == 0) /* Query - return the current rate */
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* If the requested rate is not supported the driver may just return the
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* current rate. Or preferably it may select the rate that is closest to the
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* requested one. In this sample driver we use the easy way.
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if (arg != 16000 && arg != 24000 && arg != 48000)
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* The rate was OK so write it down and report it back.
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return portc->speed = arg;
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ddksample_set_channels (int dev, short arg)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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if (arg == 0) /* Query - return the current setting */
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return portc->channels;
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* Assume 2 channels (stereo) if something incorrect is
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if (arg != 1 && arg != 2)
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return portc->channels = arg;
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ddksample_set_format (int dev, unsigned int arg)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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if (arg == 0) /* Query */
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return portc->format;
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* Fall back to 16 bits (native endianess) if some unsupported
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if (arg != AFMT_S16_NE && arg != AFMT_S32_NE)
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if (arg == AFMT_S32_NE)
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return portc->format = arg;
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ddksample_ioctl (int dev, unsigned int cmd, int *arg)
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static void ddksample_trigger (int dev, int state);
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ddksample_reset (int dev)
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ddksample_trigger (dev, 0);
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ddksample_open (int dev, int mode, int open_flags)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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ddksample_devc *devc = ossddk_adev_get_devc (dev);
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mutex_enter (&devc->mutex);
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if (portc->open_mode != 0) /* Device busy */
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mutex_exit (&devc->mutex);
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portc->open_mode = mode;
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devc->audio_open_count++;
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mutex_exit (&devc->mutex);
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ddksample_close (int dev, int mode)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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ddksample_devc *devc = ossddk_adev_get_devc (dev);
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ddksample_reset (dev);
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mutex_enter (&devc->mutex);
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portc->open_mode = 0;
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devc->audio_open_count--;
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mutex_exit (&devc->mutex);
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ddksample_trigger (int dev, int state)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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if (portc->open_mode & OPEN_WRITE)
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if ((state & PCM_ENABLE_OUTPUT)
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&& (portc->prepare_flags & PCM_ENABLE_OUTPUT))
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portc->prepare_flags &= ~PCM_ENABLE_OUTPUT;
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* Arm the timeout if it was not started yet.
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if (portc->audio_active == 0)
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timeout (ddksample_intr, portc, portc->timeout_value);
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portc->audio_active |= PCM_ENABLE_OUTPUT;
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if (!(state & PCM_ENABLE_OUTPUT)
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&& (portc->audio_active & PCM_ENABLE_OUTPUT))
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portc->audio_active &= ~PCM_ENABLE_OUTPUT;
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if (portc->audio_active == 0)
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untimeout (portc->timeout_id);
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portc->timeout_id = 0;
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if (portc->open_mode & OPEN_READ)
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if ((state & PCM_ENABLE_INPUT)
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&& (portc->prepare_flags & PCM_ENABLE_INPUT))
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portc->prepare_flags &= ~PCM_ENABLE_INPUT;
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* Arm the timeout if it was not started yet.
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if (portc->audio_active == 0)
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timeout (ddksample_intr, portc, portc->timeout_value);
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portc->audio_active |= PCM_ENABLE_INPUT;
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if (!(state & PCM_ENABLE_INPUT)
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&& (portc->audio_active & PCM_ENABLE_INPUT))
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portc->audio_active &= ~PCM_ENABLE_INPUT;
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if (portc->audio_active == 0)
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untimeout (portc->timeout_id);
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portc->timeout_id = 0;
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ddksample_prepare_for_input (int dev, int fragsize, int numfrags)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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int datarate, intrate, tmout;
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cmn_err (CE_CONT, "Input, rate=%d, channels=%d, format=0x%x, bits=%d\n",
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portc->speed, portc->channels, portc->format, portc->bits);
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* Compute the number of lbolt ticks between interrupts. This value is needed
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datarate = portc->speed * portc->channels * portc->bits / 8; /* Bytes per second */
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intrate = datarate * 10 / fragsize;
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tmout = hz * 1000 / intrate;
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cmn_err (CE_CONT, "Datarate %d, fragsize %d\n", datarate, fragsize);
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cmn_err (CE_CONT, "Intrate %d.%d interrupts / sec.\n", intrate / 10,
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cmn_err (CE_CONT, "Timeout %d.%02d ticks\n", tmout / 100, tmout % 100);
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tmout = (tmout + 50) / 100; /* Round to the nearest integer value */
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portc->timeout_value = tmout;
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portc->prepare_flags |= PCM_ENABLE_INPUT;
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ddksample_prepare_for_output (int dev, int fragsize, int numfrags)
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ddksample_portc *portc = ossddk_adev_get_portc (dev);
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int datarate, intrate, tmout;
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cmn_err (CE_CONT, "Output, rate=%d, channels=%d, format=0x%x, bits=%d\n",
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portc->speed, portc->channels, portc->format, portc->bits);
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* Compute the number of lbolt ticks between interrupts. This value is needed
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datarate = portc->speed * portc->channels * portc->bits / 8; /* Bytes per second */
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intrate = datarate * 10 / fragsize;
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tmout = hz * 1000 / intrate;
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cmn_err (CE_CONT, "Datarate %d, fragsize %d\n", datarate, fragsize);
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cmn_err (CE_CONT, "Intrate %d.%d interrupts / sec.\n", intrate / 10,
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cmn_err (CE_CONT, "Timeout %d.%02d ticks\n", tmout / 100, tmout % 100);
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tmout = (tmout + 50) / 100; /* Round to the nearest integer value */
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portc->timeout_value = tmout;
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portc->prepare_flags |= PCM_ENABLE_OUTPUT;
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ddksample_alloc_buffer (int dev, dmap_t * dmap, int direction)
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#define MY_BUFFSIZE (64*1024)
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if (ossddk_dmap_get_dmabuf (dmap) != NULL)
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ossddk_dmap_set_phys (dmap, 0); /* Not mmap() capable */
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ossddk_dmap_set_dmabuf (dmap, kmem_zalloc (MY_BUFFSIZE, KM_SLEEP));
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if (ossddk_dmap_get_dmabuf (dmap) == NULL) /* Alloc failed */
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ossddk_dmap_set_buffsize (dmap, MY_BUFFSIZE);
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ddksample_free_buffer (int dev, dmap_t * dmap, int direction)
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if (ossddk_dmap_get_dmabuf (dmap) == NULL)
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kmem_free (ossddk_dmap_get_dmabuf (dmap), MY_BUFFSIZE);
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ossddk_dmap_set_dmabuf (dmap, NULL);
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ddksample_get_buffer_pointer (int dev, dmap_t * dmap, int direction)
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static audiodrv_t ddksample_audio_driver = {
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NULL, // output_block
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ddksample_prepare_for_input,
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ddksample_prepare_for_output,
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ddksample_set_format,
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ddksample_set_channels,
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ddksample_alloc_buffer,
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ddksample_free_buffer,
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NULL /* ddksample_get_buffer_pointer */
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ddksample_audio_init (ddksample_devc * devc)
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static int rates[] = { 16000, 24000, 48000 };
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for (i = 0; i < MAX_SUBDEVICE; i++)
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ddksample_portc *portc;
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flags = ADEV_VIRTUAL | ADEV_DUPLEX;
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flags |= ADEV_SHADOW;
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if ((dev = ossddk_install_audiodev (OSS_AUDIO_DRIVER_VERSION,
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"OSS DDK sample device",
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&ddksample_audio_driver,
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AFMT_S16_NE | AFMT_S32_NE,
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portc = &devc->portc[devc->num_portc++];
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portc->speed = 48000;
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portc->format = AFMT_S16_NE;
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portc->open_mode = 0;
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portc->timeout_id = 0;
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portc->left_volume = DDKSAMPLE_MAX_VOL;
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portc->right_volume = DDKSAMPLE_MAX_VOL;
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portc->left_peak = 0;
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portc->right_peak = 0;
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ossddk_adev_set_portc (dev, portc);
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ossddk_adev_set_portnum (dev, i);
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ossddk_adev_set_mixer (dev, devc->mixer_dev);
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ossddk_adev_set_rates (dev, 16000, 48000, 3, rates);
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ossddk_adev_set_channels (dev, 1, 2); /* Mono and stereo are supported */
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* Usually the list of supported audio formats is set when calling
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* ossddk_install_audiodev(). However in some cases it may be
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* necessary to change the format list later. Changing it is
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* also necessary if recording formats are fdifferent than the
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* The next lines demonstrates how to do that. Normal drivers should
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ossddk_adev_set_formats (dev, AFMT_S32_NE | AFMT_S16_NE, // 16 and 32 bit playback
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AFMT_S16_NE); // Only 16 bit recording
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* Report stereo as the preferred channel configuration.
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ossddk_adev_set_caps (dev, DSP_CH_STEREO);
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* Many devices have some limits for the fragment size. For
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* example it's common that the fragment must fit in a single
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* memory page. In such cases setting the fragment size limits is
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* necessary. Otherwise the driver should not care about them.
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* Both the upper and lower limits are set using the same call.
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* Value of 0 means no limit.
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* Here we set the upper limit to PAGE_SIZE and no lower limit.
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ossddk_adev_set_buflimits (dev, 0, PAGE_SIZE);
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* In some cases the device may be rate clocked with some other
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* device that provides the sample clock. For example there may
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* be some special cable connected between the devices. When the
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* driver knows this it can report the device which controls
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* the sampling rate. Otherwise this information must be left
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* Knowing that two or more devices have their rates locked will
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* make work of some future OSS subsystems more reliable. However
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* false information will probably break things in the hard way.
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ossddk_adev_set_ratesource (dev, devc->some_known_device_number);
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* If the device requires some kind of proprietary control program
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* it's necessary to assign some magic number for the device.
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* The control program can find the right device by looking at
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* the magic field returned by SNDCTL_AUDIOINFO.
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* Normal devices must not report any kind of magic numbers. In
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* particular this feature must not be used to help "client"
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* programs to detect what kind of device is being used.
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* CAUTION! To avoid conflicts between two drivers using the same
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* magic it's necessary to request the magic number from
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* 4Front Technologies (support@opensound.com).
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ossddk_adev_set_magic (dev, magic_number_assigned_for_this_driver);
added
removed
misc/samples/ddksample/ddksample_audio.c
