rdpsnd_libao.c

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/* -*- c-basic-offset: 8 -*-
   rdesktop: A Remote Desktop Protocol client.
   Sound Channel Process Functions - libao-driver
   Copyright (C) Matthew Chapman 2003
   Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
   Copyright (C) Michael Gernoth mike@zerfleddert.de 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.,
   51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 This->dsp_;
ao_device *o_device = NULL;
int default_driver;
int g_samplerate;
int This->channels;
BOOL This->dsp_bu = False;
static BOOL g_reopened;
static short g_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;
    This->channels = 2;
    format.rate = 44100;
    g_samplerate = 44100;
    format.byte_format = AO_FMT_LITTLE;
 
    o_device = ao_open_live(default_driver, &format, NULL);
    if (o_device == NULL)
    {
        return False;
    }
 
    This->dsp_ = 0;
    queue_lo = queue_hi = 0;
 
    g_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;
    This->channels = pwfx->nChannels;
    format.rate = 44100;
    g_samplerate = pwfx->nSamplesPerSec;
    format.byte_format = AO_FMT_LITTLE;
 
    g_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;
    }
 
    g_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 (!This->dsp_bu)
        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 (g_reopened)
    {
        g_reopened = False;
        gettimeofday(&tv, NULL);
        prev_s = tv.tv_sec;
        prev_us = tv.tv_usec;
    }
 
    if (queue_lo == queue_hi)
    {
        This->dsp_bu = 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 (g_samplerate == 22050)
    {
        /* Resample to 44100 */
        for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_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 (This->channels == 2)
                offset = ((i * 2) - (i & 1)) * g_samplewidth;
            else
                offset = (i * 2) * g_samplewidth;
 
            memcpy(&outbuf[offset], out->p, g_samplewidth);
            memcpy(&outbuf[This->channels * g_samplewidth + offset], out->p, g_samplewidth);
 
            out->p += g_samplewidth;
            len += 2 * g_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;
    }
 
    This->dsp_bu = 1;
    return;
}