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274 lines
6.2 KiB
C
274 lines
6.2 KiB
C
/* -*- c-basic-offset: 8 -*-
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rdesktop: A Remote Desktop Protocol client.
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Sound Channel Process Functions - libao-driver
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Copyright (C) Matthew Chapman 2003
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Copyright (C) GuoJunBo guojunbo@ict.ac.cn 2003
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Copyright (C) Michael Gernoth mike@zerfleddert.de 2005
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License along
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with this program; if not, write to the Free Software Foundation, Inc.,
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51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "rdesktop.h"
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#include <unistd.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <ao/ao.h>
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#include <sys/time.h>
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#define MAX_QUEUE 10
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#define WAVEOUTBUF 16
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int This->dsp_;
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ao_device *o_device = NULL;
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int default_driver;
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int g_samplerate;
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int This->channels;
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BOOL This->dsp_bu = False;
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static BOOL g_reopened;
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static short g_samplewidth;
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static struct audio_packet
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{
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struct stream s;
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uint16 tick;
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uint8 index;
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} packet_queue[MAX_QUEUE];
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static unsigned int queue_hi, queue_lo;
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BOOL
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wave_out_open(void)
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{
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ao_sample_format format;
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ao_initialize();
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default_driver = ao_default_driver_id();
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format.bits = 16;
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format.channels = 2;
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This->channels = 2;
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format.rate = 44100;
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g_samplerate = 44100;
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format.byte_format = AO_FMT_LITTLE;
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o_device = ao_open_live(default_driver, &format, NULL);
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if (o_device == NULL)
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{
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return False;
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}
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This->dsp_ = 0;
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queue_lo = queue_hi = 0;
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g_reopened = True;
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return True;
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}
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void
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wave_out_close(void)
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{
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/* Ack all remaining packets */
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while (queue_lo != queue_hi)
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{
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rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
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free(packet_queue[queue_lo].s.data);
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queue_lo = (queue_lo + 1) % MAX_QUEUE;
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}
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if (o_device != NULL)
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ao_close(o_device);
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ao_shutdown();
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}
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BOOL
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wave_out_format_supported(WAVEFORMATEX * pwfx)
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{
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if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
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return False;
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if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
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return False;
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if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
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return False;
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/* The only common denominator between libao output drivers is a sample-rate of
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44100, we need to upsample 22050 to it */
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if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
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return False;
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return True;
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}
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BOOL
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wave_out_set_format(WAVEFORMATEX * pwfx)
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{
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ao_sample_format format;
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format.bits = pwfx->wBitsPerSample;
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format.channels = pwfx->nChannels;
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This->channels = pwfx->nChannels;
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format.rate = 44100;
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g_samplerate = pwfx->nSamplesPerSec;
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format.byte_format = AO_FMT_LITTLE;
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g_samplewidth = pwfx->wBitsPerSample / 8;
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if (o_device != NULL)
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ao_close(o_device);
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o_device = ao_open_live(default_driver, &format, NULL);
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if (o_device == NULL)
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{
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return False;
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}
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g_reopened = True;
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return True;
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}
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void
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wave_out_volume(uint16 left, uint16 right)
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{
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warning("volume changes not supported with libao-output\n");
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}
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void
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wave_out_write(STREAM s, uint16 tick, uint8 index)
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{
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struct audio_packet *packet = &packet_queue[queue_hi];
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unsigned int next_hi = (queue_hi + 1) % MAX_QUEUE;
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if (next_hi == queue_lo)
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{
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error("No space to queue audio packet\n");
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return;
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}
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queue_hi = next_hi;
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packet->s = *s;
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packet->tick = tick;
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packet->index = index;
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packet->s.p += 4;
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/* we steal the data buffer from s, give it a new one */
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s->data = malloc(s->size);
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if (!This->dsp_bu)
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wave_out_play();
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}
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void
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wave_out_play(void)
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{
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struct audio_packet *packet;
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STREAM out;
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char outbuf[WAVEOUTBUF];
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int offset, len, i;
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static long prev_s, prev_us;
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unsigned int duration;
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struct timeval tv;
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int next_tick;
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if (g_reopened)
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{
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g_reopened = False;
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gettimeofday(&tv, NULL);
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prev_s = tv.tv_sec;
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prev_us = tv.tv_usec;
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}
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if (queue_lo == queue_hi)
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{
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This->dsp_bu = 0;
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return;
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}
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packet = &packet_queue[queue_lo];
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out = &packet->s;
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if (((queue_lo + 1) % MAX_QUEUE) != queue_hi)
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{
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next_tick = packet_queue[(queue_lo + 1) % MAX_QUEUE].tick;
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}
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else
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{
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next_tick = (packet->tick + 65535) % 65536;
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}
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len = 0;
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if (g_samplerate == 22050)
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{
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/* Resample to 44100 */
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for (i = 0; (i < ((WAVEOUTBUF / 4) * (3 - g_samplewidth))) && (out->p < out->end);
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i++)
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{
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/* On a stereo-channel we must make sure that left and right
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does not get mixed up, so we need to expand the sample-
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data with channels in mind: 1234 -> 12123434
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If we have a mono-channel, we can expand the data by simply
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doubling the sample-data: 1234 -> 11223344 */
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if (This->channels == 2)
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offset = ((i * 2) - (i & 1)) * g_samplewidth;
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else
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offset = (i * 2) * g_samplewidth;
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memcpy(&outbuf[offset], out->p, g_samplewidth);
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memcpy(&outbuf[This->channels * g_samplewidth + offset], out->p, g_samplewidth);
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out->p += g_samplewidth;
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len += 2 * g_samplewidth;
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}
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}
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else
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{
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len = (WAVEOUTBUF > (out->end - out->p)) ? (out->end - out->p) : WAVEOUTBUF;
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memcpy(outbuf, out->p, len);
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out->p += len;
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}
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ao_play(o_device, outbuf, len);
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gettimeofday(&tv, NULL);
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duration = ((tv.tv_sec - prev_s) * 1000000 + (tv.tv_usec - prev_us)) / 1000;
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if (packet->tick > next_tick)
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next_tick += 65536;
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if ((out->p == out->end) || duration > next_tick - packet->tick + 500)
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{
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prev_s = tv.tv_sec;
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prev_us = tv.tv_usec;
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if (abs((next_tick - packet->tick) - duration) > 20)
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{
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DEBUG(("duration: %d, calc: %d, ", duration, next_tick - packet->tick));
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DEBUG(("last: %d, is: %d, should: %d\n", packet->tick,
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(packet->tick + duration) % 65536, next_tick % 65536));
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}
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/* Until all drivers are using the windows sound-ticks, we need to
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substract the 50 ticks added later by rdpsnd.c */
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rdpsnd_send_completion(((packet->tick + duration) % 65536) - 50, packet->index);
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free(out->data);
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queue_lo = (queue_lo + 1) % MAX_QUEUE;
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}
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This->dsp_bu = 1;
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return;
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}
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