$NetBSD: patch-ae,v 1.1.1.1 2002/09/24 12:49:13 blymn Exp $

--- src/platform/NetBSDMedia.cxx.orig	Sun Sep 22 22:29:02 2002
+++ src/platform/NetBSDMedia.cxx
@@ -0,0 +1,477 @@
+/* bzflag
+ * Copyright (c) 1993 - 2002 Tim Riker
+ *
+ * This package is free software;  you can redistribute it and/or
+ * modify it under the terms of the license found in the file
+ * named LICENSE that should have accompanied this file.
+ *
+ * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ */
+
+#include "NetBSDMedia.h"
+#include <math.h>
+#include <fcntl.h>
+#ifdef BSD
+#include <machine/endian.h>
+#else
+#ifdef __NetBSD__
+#include <sys/endian.h>
+#else
+#include <endian.h>
+#endif
+#endif
+#include <sys/time.h>
+#include <sys/types.h>
+#include <unistd.h>
+#include <sys/wait.h>
+#include "bzsignal.h"
+#ifdef __NetBSD__
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+#include <sys/ioctl.h>
+#include <TimeKeeper.h>
+#include <errno.h>
+#include <string.h>
+
+#ifdef HALF_RATE_AUDIO
+static const int defaultAudioRate=11025;
+#else
+static const int defaultAudioRate=22050;
+#endif
+
+//
+// NetBSDMedia
+//
+
+NetBSDMedia::NetBSDMedia() : BzfMedia(), audioReady(False),
+				audioPortFd(-1),
+				queueIn(-1), queueOut(-1),
+				outputBuffer(NULL),
+				childProcID(0),
+				audio8Bit(False),
+				getospaceBroken(False)
+{
+  // do nothing
+}
+
+NetBSDMedia::~NetBSDMedia()
+{
+  // do nothing
+}
+
+double			NetBSDMedia::getTime()
+{
+  struct timeval tv;
+  gettimeofday(&tv, 0);
+  return (double)tv.tv_sec + 1.0e-6 * (double)tv.tv_usec;
+}
+
+double			NetBSDMedia::stopwatch(boolean start)
+{
+  if (start) {
+    stopwatchTime = getTime();
+    return 0.0;
+  }
+  return getTime() - stopwatchTime;
+}
+
+void			NetBSDMedia::sleep(float timeInSeconds)
+{
+  struct timeval tv;
+  tv.tv_sec = (long)timeInSeconds;
+  tv.tv_usec = (long)(1.0e6 * (timeInSeconds - tv.tv_sec));
+  select(0, NULL, NULL, NULL, &tv);
+}
+
+boolean			NetBSDMedia::openAudio()
+{
+  // don't re-initialize
+  if (audioReady) return False;
+
+  // check for and open audio hardware
+  if (!checkForAudioHardware() || !openAudioHardware()) return False;
+
+  // open communication channel (FIFO pipe).  close on exec.
+  int fd[2];
+  if (pipe(fd)<0) {
+    closeAudio();
+    return False;
+  }
+  queueIn = fd[1];
+  queueOut = fd[0];
+  fcntl(queueIn,  F_SETFL, fcntl(queueIn, F_GETFL, 0)  | O_NDELAY);
+  fcntl(queueOut, F_SETFL, fcntl(queueOut, F_GETFL, 0) | O_NDELAY);
+  fcntl(queueIn,  F_SETFD, fcntl(queueIn, F_GETFD, 0)  | FD_CLOEXEC);
+  fcntl(queueOut, F_SETFD, fcntl(queueOut, F_GETFD, 0) | FD_CLOEXEC);
+
+  // compute maxFd for use in select() call
+  maxFd = queueOut;
+  if (maxFd<audioPortFd) maxFd = audioPortFd;
+  maxFd++;
+
+  // make an output buffer
+  outputBuffer = new short[audioBufferSize];
+
+  // Set default no thread
+  childProcID=0;
+
+  // ready to go
+  audioReady = True;
+  return True;
+}
+
+boolean			NetBSDMedia::checkForAudioHardware()
+{
+  boolean flag=False;
+  if (!access("/dev/audio", W_OK)) flag=True;
+  if (!access("/dev/sound", W_OK)) flag=True;
+  return flag;
+}
+
+boolean			NetBSDMedia::openIoctl(
+				int cmd, void* value, boolean req)
+{
+  if (audioPortFd == -1)
+      return False;
+
+  if (ioctl(audioPortFd, cmd, value) < 0) {
+    fprintf(stderr, "audio ioctl failed (cmd %x, err %d)... ", cmd, errno);
+    if (req) {
+      close(audioPortFd);
+      audioPortFd = -1;
+      fprintf(stderr, "giving up on audio\n");
+    }
+    else {
+      fprintf(stderr, "ignored\n");
+    }
+    return False;
+  }
+  return True;
+}
+
+static const int	NumChunks = 4;
+
+boolean			NetBSDMedia::openAudioHardware()
+{
+  int format, n;
+
+  // what's the audio format?
+#if BYTE_ORDER == BIG_ENDIAN
+  format = AFMT_S16_BE;
+#else
+  format = AFMT_S16_LE;
+#endif
+
+  // what the frequency?
+  audioOutputRate = defaultAudioRate;
+
+  // how big a fragment to use?  we want to hold at around 1/10th of
+  // a second.
+  int fragmentSize = (int)(0.08f * (float)audioOutputRate);
+  n = 0;
+  while ((1 << n) < fragmentSize)
+    ++n;
+
+  // samples are two bytes each and we're in stereo so quadruple the size
+  fragmentSize = n + 2;
+
+  // now how many fragments and what's the low water mark (in fragments)?
+  int fragmentInfo = (NumChunks << 16) | fragmentSize;
+  audioLowWaterMark = 2;
+
+  // open device (but don't wait for it)
+  audioPortFd = open("/dev/audio", O_WRONLY | O_NDELAY, 0);
+  if (audioPortFd == -1) {
+    audioPortFd = open("/dev/sound", O_WRONLY | O_NDELAY, 0);
+    if (audioPortFd == -1) {
+      fprintf(stderr, "Failed to open audio device /dev/audio or /dev/sound (%d)\n", errno);
+      return False;
+	}
+  }
+
+  // back to blocking I/O
+  fcntl(audioPortFd, F_SETFL, fcntl(audioPortFd, F_GETFL, 0) & ~O_NDELAY);
+
+  /* close audio on exec so launched server doesn't hold sound device */
+  fcntl(audioPortFd, F_SETFD, fcntl(audioPortFd, F_GETFD) | FD_CLOEXEC);
+
+  // initialize device
+  openIoctl(SNDCTL_DSP_RESET, 0);
+  n = fragmentInfo;
+  noSetFragment = false;
+  if (!openIoctl(SNDCTL_DSP_SETFRAGMENT, &n, False)) {
+    // this is not good.  we can't set the size of the fragment
+    // buffers.  we'd like something short to minimize latencies
+    // and the default is probably too long.  we've got two
+    // options here:  accept the latency or try to force the
+    // driver to play partial fragments.  we'll try the later
+    // unless BZF_AUDIO_NOPOST is in the environment
+    if (!getenv("BZF_AUDIO_NOPOST"))
+      noSetFragment = true;
+  }
+  n = format;
+  openIoctl(SNDCTL_DSP_SETFMT, &n, False);
+  if (n != format) {
+    audio8Bit = True;
+    n = AFMT_U8;
+    openIoctl(SNDCTL_DSP_SETFMT, &n);
+  }
+  n = 1;
+  openIoctl(SNDCTL_DSP_STEREO, &n);
+  n = defaultAudioRate;
+  openIoctl(SNDCTL_DSP_SPEED, &n);
+
+  // set audioBufferSize, which is the number of samples (not bytes)
+  // in each fragment.  there are two bytes per sample so divide the
+  // fragment size by two unless we're in audio8Bit mode.  also, if
+  // we couldn't set the fragment size then force the buffer size to
+  // the size we would've asked for.  we'll force the buffer to be
+  // flushed after we write that much data to keep latency low.
+  if (noSetFragment ||
+	!openIoctl(SNDCTL_DSP_GETBLKSIZE, &audioBufferSize, False) ||
+	audioBufferSize > (1 << fragmentSize)) {
+    audioBufferSize = 1 << fragmentSize;
+    noSetFragment = true;
+  }
+  if (!audio8Bit)
+    audioBufferSize >>= 1;
+
+  // SNDCTL_DSP_GETOSPACE not supported on all platforms.  check if
+  // it fails here and, if so, do a workaround by using the wall
+  // clock.  *shudder*
+  if (audioPortFd != -1) {
+    audio_buf_info info;
+    if (!openIoctl(SNDCTL_DSP_GETOSPACE, &info, False)) {
+      getospaceBroken = True;
+      chunksPending = 0;
+      chunksPerSecond = (double)getAudioOutputRate() /
+				(double)getAudioBufferChunkSize();
+    }
+  }
+
+  return (audioPortFd != -1);
+}
+
+void			NetBSDMedia::closeAudio()
+{
+  delete [] outputBuffer;
+  if (audioPortFd>=0) close(audioPortFd);
+  if (queueIn!=-1) close(queueIn);
+  if (queueOut!=-1) close(queueOut);
+  audioReady=False;
+  audioPortFd=-1;
+  queueIn=-1;
+  queueOut=-1;
+  outputBuffer=0;
+}
+
+boolean			NetBSDMedia::startAudioThread(
+				void (*proc)(void*), void* data)
+{
+  // if no audio thread then just call proc and return
+  if (!hasAudioThread()) {
+    proc(data);
+    return True;
+  }
+
+  // has an audio thread so fork and call proc
+  if (childProcID) return True;
+  if ((childProcID=fork()) > 0) {
+    close(queueOut);
+    close(audioPortFd);
+    return True;
+  }
+  else if (childProcID < 0) {
+    return False;
+  }
+  close(queueIn);
+  proc(data);
+  exit(0);
+}
+
+void			NetBSDMedia::stopAudioThread()
+{
+  if (childProcID != 0) kill(childProcID, SIGTERM);
+  childProcID=0;
+}
+
+boolean			NetBSDMedia::hasAudioThread() const
+{
+#if defined(NO_AUDIO_THREAD)
+  return False;
+#else
+  return True;
+#endif
+}
+
+void			NetBSDMedia::audioThreadInit(void*)
+{
+}
+
+void			NetBSDMedia::writeSoundCommand(const void* cmd, int len)
+{
+  if (!audioReady) return;
+  write(queueIn, cmd, len);
+}
+
+boolean			NetBSDMedia::readSoundCommand(void* cmd, int len)
+{
+  return (read(queueOut, cmd, len)==len);
+}
+
+int			NetBSDMedia::getAudioOutputRate() const
+{
+  return audioOutputRate;
+}
+
+int			NetBSDMedia::getAudioBufferSize() const
+{
+  return NumChunks*(audioBufferSize>>1);
+}
+
+int			NetBSDMedia::getAudioBufferChunkSize() const
+{
+  return audioBufferSize>>1;
+}
+
+boolean			NetBSDMedia::isAudioTooEmpty() const
+{
+  if (getospaceBroken) {
+    if (chunksPending > 0) {
+      // get time elapsed since chunkTime
+      const double dt = getTime() - chunkTime;
+
+      // how many chunks could've played in the elapsed time?
+      const int numChunks = (int)(dt * chunksPerSecond);
+
+      // remove pending chunks
+      NetBSDMedia* self = (NetBSDMedia*)this;
+      self->chunksPending -= numChunks;
+      if (chunksPending < 0)
+	self->chunksPending = 0;
+      else
+	self->chunkTime += (double)numChunks / chunksPerSecond;
+    }
+    return chunksPending < audioLowWaterMark;
+  }
+  else {
+    audio_buf_info info;
+    if (ioctl(audioPortFd, SNDCTL_DSP_GETOSPACE, &info) < 0)
+      return False;
+    return info.fragments > info.fragstotal - audioLowWaterMark;
+  }
+}
+
+void			NetBSDMedia::writeAudioFrames8Bit(
+				const float* samples, int numFrames)
+{
+  int numSamples = 2 * numFrames;
+  int limit;
+  char *smOutputBuffer;
+
+  smOutputBuffer=(char*)outputBuffer;
+  while (numSamples > 0) {
+    if (numSamples>audioBufferSize) limit=audioBufferSize;
+    else limit=numSamples;
+    for (int j = 0; j < limit; j++) {
+      if (samples[j] <= -32767.0) smOutputBuffer[j] = 0;
+      else if (samples[j] >= 32767.0) smOutputBuffer[j] = 255;
+      else smOutputBuffer[j] = char((samples[j]+32767)/257);
+    }
+
+    // fill out the chunk (we never write a partial chunk)
+    if (limit < audioBufferSize) {
+      for (int j = limit; j < audioBufferSize; ++j)
+	smOutputBuffer[j] = 127;
+      limit = audioBufferSize;
+    }
+
+    write(audioPortFd, smOutputBuffer, limit);
+    samples += limit;
+    numSamples -= limit;
+  }
+}
+
+void			NetBSDMedia::writeAudioFrames16Bit(
+				const float* samples, int numFrames)
+{
+  int numSamples = 2 * numFrames;
+  int limit;
+
+  while (numSamples > 0) {
+    if (numSamples>audioBufferSize) limit=audioBufferSize;
+    else limit=numSamples;
+    for (int j = 0; j < limit; j++) {
+      if (samples[j] < -32767.0) outputBuffer[j] = -32767;
+      else if (samples[j] > 32767.0) outputBuffer[j] = 32767;
+      else outputBuffer[j] = short(samples[j]);
+    }
+
+    // fill out the chunk (we never write a partial chunk)
+    if (limit < audioBufferSize) {
+      for (int j = limit; j < audioBufferSize; ++j)
+	outputBuffer[j] = 0;
+      limit = audioBufferSize;
+    }
+
+    write(audioPortFd, outputBuffer, 2*limit);
+    samples += limit;
+    numSamples -= limit;
+  }
+}
+
+void			NetBSDMedia::writeAudioFrames(
+				const float* samples, int numFrames)
+{
+  if (audio8Bit) writeAudioFrames8Bit(samples, numFrames);
+  else writeAudioFrames16Bit(samples, numFrames);
+
+  // if we couldn't set the fragment size then force the driver
+  // to play the short buffer.
+  if (noSetFragment) {
+    int dummy = 0;
+    ioctl(audioPortFd, SNDCTL_DSP_POST, &dummy);
+  }
+
+  if (getospaceBroken) {
+    if (chunksPending == 0)
+      chunkTime = getTime();
+    chunksPending += (numFrames + getAudioBufferChunkSize() - 1) /
+						getAudioBufferChunkSize();
+  }
+}
+
+void			NetBSDMedia::audioSleep(
+				boolean checkLowWater, double endTime)
+{
+  fd_set commandSelectSet;
+  struct timeval tv;
+
+  // To do both these operations at once, we need to poll.
+  if (checkLowWater) {
+    // start looping
+    TimeKeeper start = TimeKeeper::getCurrent();
+    do {
+      // break if buffer has drained enough
+      if (isAudioTooEmpty()) break;
+      FD_ZERO(&commandSelectSet);
+      FD_SET(queueOut, &commandSelectSet);
+      tv.tv_sec=0;
+      tv.tv_usec=50000;
+      if (select(maxFd, &commandSelectSet, 0, 0, &tv)) break;
+
+    } while (endTime<0.0 || (TimeKeeper::getCurrent()-start)<endTime);
+  } else {
+    FD_ZERO(&commandSelectSet);
+    FD_SET(queueOut, &commandSelectSet);
+    tv.tv_sec=int(endTime);
+    tv.tv_usec=int(1.0e6*(endTime-floor(endTime)));
+
+    select(maxFd, &commandSelectSet, 0, 0, (endTime>=0.0)?&tv : 0);
+  }
+}
+// ex: shiftwidth=2 tabstop=8