source: vbox/trunk/src/VBox/RDP/client/rdpsnd_libao.c@ 8973

/* -*- c-basic-offset: 8 -*-
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - libao-driver
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo [email protected] 2003
   Copyright (C) Michael Gernoth [email protected] 2005

   This program is free software; you can redistribute it and/or modify
   it under the terms of the GNU General Public License as published by
   the Free Software Foundation; either version 2 of the License, or
   (at your option) any later version.

   This program is distributed in the hope that it will be useful,
   but WITHOUT ANY WARRANTY; without even the implied warranty of
   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
   GNU General Public License for more details.

   You should have received a copy of the GNU General Public License
   along with this program; if not, write to the Free Software
   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/

#include "rdesktop.h"
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <ao/ao.h>
#include <sys/time.h>

#define MAX_QUEUE       10
#define WAVEOUTBUF      16

int g_dsp_fd;
BOOL g_dsp_busy = False;
static ao_device *o_device = NULL;
static int default_driver;
static int samplerate;
static int audiochannels;
static BOOL reopened;
static short samplewidth;

static struct audio_packet
{
        struct stream s;
        uint16 tick;
        uint8 index;
} packet_queue[MAX_QUEUE];
static unsigned int queue_hi, queue_lo;

BOOL
wave_out_open(void)
{
        ao_sample_format format;

        ao_initialize();
        default_driver = ao_default_driver_id();

        format.bits = 16;
        format.channels = 2;
        audiochannels = 2;
        format.rate = 44100;
        samplerate = 44100;
        format.byte_format = AO_FMT_LITTLE;

        o_device = ao_open_live(default_driver, &format, NULL);
        if (o_device == NULL)
        {
                return False;
        }

        g_dsp_fd = 0;
        queue_lo = queue_hi = 0;

        reopened = True;

        return True;
}

void
wave_out_close(void)
{
        /* Ack all remaining packets */
        while (queue_lo != queue_hi)
        {
                rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
                free(packet_queue[queue_lo].s.data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        if (o_device != NULL)
                ao_close(o_device);

        ao_shutdown();
}

BOOL
wave_out_format_supported(WAVEFORMATEX * pwfx)
{
        if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
                return False;
        if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
                return False;
        if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
                return False;
        /* The only common denominator between libao output drivers is a sample-rate of
           44100, we need to upsample 22050 to it */
        if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
                return False;

        return True;
}

BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
        ao_sample_format format;

        format.bits = pwfx->wBitsPerSample;
        format.channels = pwfx->nChannels;
        audiochannels = pwfx->nChannels;
        format.rate = 44100;
        samplerate = pwfx->nSamplesPerSec;
        format.byte_format = AO_FMT_LITTLE;

        samplewidth = pwfx->wBitsPerSample / 8;

        if (o_device != NULL)
                ao_close(o_device);

        o_device = ao_open_live(default_driver, &format, NULL);
        if (o_device == NULL)
        {
                return False;
        }

        reopened = True;

        return True;
}

void
wave_out_volume(uint16 left, uint16 right)
{
        warning("volume changes not supported with libao-output\n");
}

void
wave_out_write(STREAM s, uint16 tick, uint8 index)
{
        struct audio_packet *packet = &packet_queue[queue_hi];
        unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;

        if (next_hi == queue_lo)
        {
                error("No space to queue audio packet\n");
                return;
        }

        queue_hi = next_hi;

        packet->s = *s;
        packet->tick = tick;
        packet->index = index;
        packet->s.p += 4;

        /* we steal the data buffer from s, give it a new one */
        s->data = malloc(s->size);

        if (!g_dsp_busy)
                wave_out_play();
}

void
wave_out_play(void)
{
        struct audio_packet *packet;
        STREAM out;
        char outbuf[WAVEOUTBUF];
        int offset, len, i;
        static long prev_s, prev_us;
        unsigned int duration;
        struct timeval tv;
        int next_tick;

        if (reopened)
        {
                reopened = False;
                gettimeofday(&tv, NULL);
                prev_s = tv.tv_sec;
                prev_us = tv.tv_usec;
        }

        if (queue_lo == queue_hi)
        {
                g_dsp_busy = 0;
                return;
        }

        packet = &packet_queue[queue_lo];
        out = &packet->s;

        if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
        {
                next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
        }
        else
        {
                next_tick = (packet->tick + 65535) % 65536;
        }

        len = 0;

        if (samplerate == 22050)
        {
                /* Resample to 44100 */
                for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - samplewidth))) && (out->p < out->end);
                     i++)
                {
                        /* On a stereo-channel we must make sure that left and right
                           does not get mixed up, so we need to expand the sample-
                           data with channels in mind: 1234 -> 12123434
                           If we have a mono-channel, we can expand the data by simply
                           doubling the sample-data: 1234 -> 11223344 */
                        if (audiochannels == 2)
                                offset = ((i * 2) - (i & 1)) * samplewidth;
                        else
                                offset = (i * 2) * samplewidth;

                        memcpy(&outbuf[offset], out->p, samplewidth);
                        memcpy(&outbuf[audiochannels * samplewidth + offset], out->p, samplewidth);

                        out->p += samplewidth;
                        len += 2 * samplewidth;
                }
        }
        else
        {
                len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
                memcpy(outbuf, out->p, len);
                out->p += len;
        }

        ao_play(o_device, outbuf, len);

        gettimeofday(&tv, NULL);

        duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;

        if (packet->tick > next_tick)
                next_tick += 65536;

        if ((out->p == out->end) || duration > next_tick - packet->tick + 500)
        {
                prev_s = tv.tv_sec;
                prev_us = tv.tv_usec;

                if (abs((next_tick - packet->tick) - duration) > 20)
                {
                        DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
                        DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
                               (packet->tick + duration) % 65536, next_tick % 65536));
                }

                /* Until all drivers are using the windows sound-ticks, we need to
                   substract the 50 ticks added later by rdpsnd.c */
                rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
                free(out->data);
                queue_lo = (queue_lo + 1) % MAX_QUEUE;
        }

        g_dsp_busy = 1;
        return;
}