/*
 * Copyright (c) 1991-1995 Regents of the University of California.
 * All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 * 3. All advertising materials mentioning features or use of this software
 *    must display the following acknowledgement:
 *	This product includes software developed by the Computer Systems
 *	Engineering Group at Lawrence Berkeley Laboratory.
 * 4. Neither the name of the University nor of the Laboratory may be used
 *    to endorse or promote products derived from this software without
 *    specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
 * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
 * SUCH DAMAGE.
 */
static const char rcsid[] =
    "@(#) $Header: /work/projects/tove/cvs/src/testing/vat/controller.cc,v 1.1 1997/12/08 17:22:50 parnanen Exp $ (LBL)";

#define AUDIO_SPS 8000	/* audio samples per second (used to convert
			 * playout delay from seconds to bytes of buffer).
			 * This is the sample rate used by the audio
			 * hardware.  If this rate is different than
			 * the sample rate of some network audio format,
			 * sample rate conversion has to be done between
			 * the encoder/decoder objects & here.  Note
			 * that sample rate conversion is a very compute
			 * intensive operation & there isn't currently
			 * any support for it in vat.  Also note that
			 * there are many control variables expressed
			 * in terms of a 160 sample (20ms at 8KHz)
			 * audio frame size and some of these need to
			 * be changed if the AUDIO_SPS is changed.
			 */
#define SS_GRANULARITY 1440	/* sample stream sizes are always rounded
				 * to some integer multiple of this number.
				 * It should be set to the least common
				 * multiple of the possible *output* audio
				 * frame sizes (see comments below).
				 * In our case, possible frame sizes are
				 * 160 & 180 samples.
				 */
/*
 * following is timeout interval (in ms) when vat does not have
 * the audio & is running off a system timer instead.  This interval
 * must be the same as the audio read blocksize (FRAMESIZE /
 * AUDIO_SPS * 1000 ms) and must be < 1 sec.
 */
#define FRAME_TIME (FRAMESIZE * 1000 / AUDIO_SPS)

#include "config.h"
#include "sys-time.h"
#include "audio.h"
#include "ss.h"
#include "controller.h"
#include "encoder.h"
#include "mulaw.h"
#include "vu.h"
#include "Tcl.h"
#include "ntp-time.h"

#define METER_UPDATE_FREQ 3

/*
 * A controller class specifically for handling half-duplex audio
 * devices like those commonly found in PCs.  It derives its timebase
 * from a 20ms system timer rather than using audio read completions
 * like the normal controller class.
 */
class HDController : public Controller {
    public:
	HDController();
	virtual void update(Observable*);
    protected:
	virtual void timeout();
	virtual void audio_handle();
};

static class ControllerMatch : public Matcher {
    public:
	ControllerMatch() : Matcher("controller") {}
	TclObject* match(const char* id) {
		if (strcasecmp(id, "full-duplex") == 0)
			return (new Controller);
		else if (strcasecmp(id, "half-duplex") == 0)
			return (new HDController);
		else
			return (0);
	}
} controller_matcher;


Controller::Controller() :
	audio_(0),
	lastnetout_(0),
	lastaudout_(0),
#ifdef WIN32
	/* windows is very, very slow & can't hack 20ms events */
	timer_interval_(FRAME_TIME * 4),
#else
	timer_interval_(FRAME_TIME),
#endif
	tsec_(0),
	tusec_(0),
	ostate_(TALK_TAIL + TALK_LEAD),
	pmeter_(0),
	rmeter_(0),
	talk_thresh_(0),
	echo_thresh_(0),
	echo_suppress_time_(0),
	meter_update_(METER_UPDATE_FREQ),
	active_(0),
	encoder_(0),
	outmax_(FRAMESIZE),
	outlen_(0),
	out_ts_(0),
	out_(0)
{
	Tcl& tcl = Tcl::instance();
	echo_thresh_ = atoi(tcl.attr("echoThresh"));
	echo_suppress_time_ = atoi(tcl.attr("echoSuppressTime")) / 20 * FRAMESIZE;
	idle_drop_time_ = atoi(tcl.attr("idleDropTime")) * 50;

	int maxplayout = atoi(tcl.attr("maxPlayout")) * AUDIO_SPS;
	if (maxplayout < (TALK_LEAD + TALK_TAIL + 2) * FRAMESIZE) {
		printf(" max playout delay %d too short - using %d sec.\n",
			maxplayout,
			((TALK_LEAD + TALK_TAIL + 2) * FRAMESIZE +
			AUDIO_SPS - 1) / AUDIO_SPS);
		maxplayout = (TALK_LEAD + TALK_TAIL + 2) * FRAMESIZE;
	}
	maxplayout = (maxplayout + (SS_GRANULARITY - 1)) / SS_GRANULARITY;
	maxplayout *= SS_GRANULARITY / FRAMESIZE;

	/*XXX tk script */
	int magclevel = atoi(tcl.attr("mikeAGCLevel"));
	int sagclevel = atoi(tcl.attr("speakerAGCLevel"));

	as_ = new SampleStream(FRAMESIZE, maxplayout,
			       (TALK_LEAD+1)*FRAMESIZE, magclevel);
	ns_ = new SampleStream(FRAMESIZE, maxplayout,
			       (TALK_LEAD+1)*FRAMESIZE, sagclevel);

	lastaudout_ = as_->Clock();
	lastnetout_ = ns_->Clock();

	int thresh = atoi(tcl.attr("silenceThresh"));
	as_->ssthresh(thresh);
	ns_->ssthresh(thresh);
}

void Controller::update(Observable*)
{
	if (!audio_->HaveAudio()) {
		timeval tv;
		::gettimeofday(&tv, 0);
		tsec_ = tv.tv_sec;
		tusec_ = tv.tv_usec;
		/* Reset the meters and force a redraw. */
		if (pmeter_ != 0) {
			pmeter_->set(0.); pmeter_->redraw();
			rmeter_->set(0.); rmeter_->redraw();
		}
		msched(timer_interval_);
	} else {
		cancel();
	}
}

void Controller::DoAudio()
{
	double rlevel, plevel;
	register int nsmean = ns_->LTMean();

	if (ns_->Max())
		lastnetout_ = as_->Clock();
	ns_->Compute();
	plevel = ns_->Mean();
	mixaudio(*ns_);
	ns_->UpdateAGC();

	if (audio_->RMuted()) {
		rlevel = 0.;
	} else {
		as_->Compute();
		rlevel = as_->Mean();
		/*
		 * Next check is to cut residual echo in speakerphone &
		 * echo-cancel modes: If we haven't been talking recently, have
		 * just sent data to the speaker and the signal we got from the
		 * mike was about the same as the signal we played, assume the
		 * mike signal is echo.  NOTE that nsmean was grabbed before
		 * ns_->Compute() so we are comparing the previous audio output
		 * to the current audio input - this should make it more likely
		 * that we're looking at the echo signal (though under Sun OS
		 * there's still a lot of buffer in the STREAM I/O system).
		 */
		if (audio_->Mode() != Audio::mode_none && !sending() &&
		    (as_->LTMean() - nsmean < echo_thresh_ ||
		     as_->Silent(talk_thresh_)))
			;
		else if (audio_->Mode() != Audio::mode_netmutesmike ||
			 lastnetout_ == 0 ||
			 u_int(as_->Clock() - lastnetout_) > echo_suppress_time_) {
			if (Output()) {
				lastnetout_ = 0;
				active_ = 1;
				as_->UpdateAGC();
			}
		}
	}
	/*
	 * Update the meters.  We control the rate with METER_UPDATE_FREQ
	 * to cut down on CPU load from the X window updates.
	 */
	if (pmeter_ != 0 && --meter_update_ <= 0) {
		pmeter_->set(plevel);
		rmeter_->set(rlevel);
		meter_update_ = METER_UPDATE_FREQ;
	}
}

void Controller::DoTimer()
{
	/* Advance to next audio frame */
	as_->Advance();
	ns_->Advance();
}

/*
 * Called from decoders to mix in the block of samples 'del'
 * samples into the future.
 */
void Controller::mix_from_net(int del, const u_int8_t* frame, int len)
{
	if (del >= 0)
		ns_->Mix(del, frame, len);
}

void Controller::audio_handle()
{
	DoAudio();
	DoTimer();
}

/*
 * Called when we don't have the audio device.  Normally, our time
 * base comes from the audio device's sample clock, so when we don't
 * have the device open, we revert to timers.
 */
void Controller::timeout()
{
	/*
	 * Use get time of day and keep track of the current hard time
	 * in the tsec_/tusec_ variables.  Tk timers are an unreliable
	 * time base.  We use them to dispatch an event here then call
	 * gettimeofday to see how many times we really should have
	 * been called in the intervening period.
	 */
	timeval tv;
	::gettimeofday(&tv, 0);
	u_int u = (u_int)tv.tv_usec;
	u_int s = (u_int)tv.tv_sec;
	if (s > tsec_ + 3) {
		/*
		 * We're way behind.  Most likely we were suspended and
		 * then resumed.  Instead of trying to catch up, just resync.
		 */
		tusec_ = u;
		tsec_ = s;
	}
	while ((int(tusec_ - u) <= 0 && s == tsec_) || int(tsec_ - s) < 0) {
		DoTimer();
		while ((tusec_ += 1000 * FRAME_TIME) >= 1000000) {
			tusec_ -= 1000000;
			++tsec_;
		}
	}
	msched(timer_interval_);
}

/*
 * send the next block of samples to the encoder.  We might coalesce
 * several calls into a larger chunk (depending on the value of outmax_).
 * The point is to stuff more data into a single packet to amortize the
 * packet header overhead (at the cost of increased latency).
 */
void Controller::send_block(u_int32_t ts, u_int8_t* blk, int len)
{
	if (out_ == 0) {
		out_ = blk;
		outlen_ = 0;
		out_ts_ = ts;
	} else if (&out_[outlen_] != blk) {
		/*
		 * frames wrapped in ss buffer -- copy to buf to
		 * keep things contiguous.
		 */
		if (out_ != overflow_) {
			/* first time */
			memcpy(overflow_, out_, outlen_);
			out_ = overflow_;
		}
		/* copy current chunk */
		memcpy(&out_[outlen_], blk, len);
	}
	outlen_ += len;
	if (outlen_ >= outmax_) {
		encoder_->encode(out_ts_, out_, outlen_);
		out_ = 0;
	}
}

/*
 * get the current media timestamp
 */
u_int32_t Controller::media_ts()
{
	::gettimeofday(&last_uts_, 0);
	u_int32_t ts = as_->Clock();
	last_mts_ = ts;
	return (ts);
}

u_int32_t Controller::ref_ts()
{
	timeval now;
	::gettimeofday(&now, 0);
	int t = (now.tv_sec - last_uts_.tv_sec) * 8000;
	t += ((now.tv_usec - last_uts_.tv_usec) << 3) / 1000;
	return (last_mts_ + t);
}

/*
 * Check for silence and otherwise send audio frames to the encoder.
 * If this is the start of a talk-spurt, go back TALK_LEAD blocks into
 * the past and send them too (because the silence detector isn't
 * perfect especially near a silence-to-speech transition).  At the
 * end of a talk-spurt, send TALK_TAIL extra blocks because the
 * silence detector is unreliable near a speech-to-silence transition.
 */
int Controller::Output()
{
	register int bs = as_->BlkSize();
	if (as_->Silent()) {
		if (ostate_ >= TALK_TAIL) {
			/* between talk spurts */
			if (out_ != 0) {
				/* flush the partial last block */
				send_block(media_ts(), as_->BlkBack(0), bs);
				return (1);
			}
			if (ostate_ < TALK_TAIL+TALK_LEAD)
				++ostate_;
			return (0);
		}
		++ostate_;
	} else if (ostate_) {
		/*
		 * if start of talk after silence, generate packets for
		 * any leading speech that we might have missed.
		 */
		for (int i = (ostate_ - TALK_TAIL) * bs; i > 0; i -= bs) {
			u_int32_t ts = as_->Clock() - i;
			send_block(ts, as_->BlkBack(i), bs);
		}
		ostate_ = 0;
	}
	send_block(media_ts(), as_->BlkBack(0), bs);
	return (1);
}

extern "C" {
extern u_char tonemax[];
extern u_char tone0dBm[];
extern u_char tone6dBm[];
}

void Controller::mixaudio(SampleStream& ss)
{
	/*
	 * The audio driver is ready to give us the next packet.
	 * This serves as our time base.  We do the following:
	 *  - output next chunk to audio driver.
	 *    (We do the output *first* so the rest of our
	 *    processing is overlapped with the real-time audio
	 *    output, otherwise we tend to accumulate estimate
	 *    random delays.)
	 *  - If we're doing echo cancellation, estimate the
	 *    echo resulting from the block just output and mix
	 *    the echo inverse into the *input* sample stream
	 *    at the estimated echo delay.
	 *  - read packet from audio
	 *  - Mix the input data into the input sample stream
	 *    (this is a mix because we might have an inverse
	 *    echo signal or tone that we want summed with the
	 *    input).
	 */
	u_char* blk = 0;

	int loopback = audio_->GetLoopback();
	/*XXX*/
#ifdef notdef
	int blksize = audio_->BlockSize();
#else
	int blksize = as_->BlkSize();
#endif

	if (loopback != Audio::loop_none) {
		blk = audio_->Read();
		switch (loopback) {
		case Audio::loop_none:
			break;
		case Audio::loop_mike:
			ss.Mix(0, blk, blksize);
			break;
		case Audio::loop_tone6:
			ss.Mix(0, tone6dBm, blksize);
			break;
		case Audio::loop_tone0:
			ss.Mix(0, tone0dBm, blksize);
			break;
		case Audio::loop_tonemax:
			ss.Mix(0, tonemax, blksize);
			break;
		}
	}

	/*
	 * Now write a block of samples to the audio device provided
	 * the following conditions hold:
	 *
	 * (1)	the audio output isn't muted
	 *
	 * (2)	there won't be an echo problem i.e., we're not sending, OR
	 * 	we're in not in the mode where input has priority over output,
	 *	OR the mike is muted.  this last check is not strictly
	 *	necessary but allows the user to immediately hear the
	 *	far end after muting the mike (rather than waiting for
	 *	TALK_TAIL extra audio blocks to drain).
	 *
	 * (3)	the signal to output isn't slience, or we're running
	 *	a loopback test.  i.e., if we're about to write a
	 *	completely silent frame (i.e., no packets at all from
	 *	the network) then don't do it.  This prevents a backlog
	 *	of samples (i.e., a net delay) to build up in the audio
	 *	driver.  Note that this problem is completely independent
	 *	of the silent suppression solution for the outbound path.
	 */
	if (! audio_->PMuted() &&
	    (!sending() || audio_->Mode() != Audio::mode_mikemutesnet ||
	     audio_->RMuted()) &&
	    (loopback || (ss.Max() != 0 && ss.LTMean() != 0))) {
		/*
		 * if we haven't written for a while,
		 * write an extra block to generate a bit
		 * of a backlog between us & the driver.
		 */
		u_int32_t sc = as_->Clock();
		if (u_int(sc - lastaudout_) > 4*FRAMESIZE)
			audio_->Write(ss.BlkBack(blksize));
		audio_->Write(ss.CurBlk());
		lastaudout_ = sc;
		active_ = 1;
#ifdef notyet
		if (mode == mode_ec && !rmute) {
			int offset = AdjustTime(0);
			u_char ecblk[MAXAUDIOSIZE+4];
			int resid = offset & 3;
			if (resid)
				ecblk[0] = 0x7f7f7f7f;
			filter->Compute(os, &ecblk[resid],
					blksize);
			as->Mix(offset &~ 3, ecblk, blksize);
		}
#endif
	}
	if (loopback == Audio::loop_none)
		blk = audio_->Read();

	as_->Mix(0, blk, blksize);
}


HDController::HDController()
{
	timeval tv;
	::gettimeofday(&tv, 0);
	tsec_ = tv.tv_sec;
	tusec_ = tv.tv_usec;
	msched(timer_interval_);
}

void HDController::update(Observable*)
{
	/*
	 * do nothing -- leave the timer running even when
	 * we have the audio
	 */
}

void HDController::timeout()
{
	timeval tv;
	::gettimeofday(&tv, 0);
	u_int u = (u_int)tv.tv_usec;
	u_int s = (u_int)tv.tv_sec;
	if (s > tsec_ + 3) {
		/*
		 * We're way behind.  Most likely we were suspended and
		 * then resumed.  Instead of trying to catch up, just resync.
		 */
		tusec_ = u;
		tsec_ = s;
	}
	while ((int(tusec_ - u) <= 0 && s == tsec_) || int(tsec_ - s) < 0) {
		DoTimer();
		if (audio_->HaveAudio() && audio_->FrameReady())
			DoAudio();

		while ((tusec_ += 1000 * FRAME_TIME) >= 1000000) {
			tusec_ -= 1000000;
			++tsec_;
		}
	}
	msched(timer_interval_);
}

void HDController::audio_handle()
{
	printf("HDController::audio_handle()\n");
}

int Controller::command(int argc, const char*const* argv)
{
	Tcl& tcl = Tcl::instance();
	if (argc == 2) {
		if (strcmp(argv[1], "ntp-time") == 0) {
			sprintf(tcl.buffer(), "%u", ntptime());
			tcl.result(tcl.buffer());
			return (TCL_OK);
		}
		if (strcmp(argv[1], "unix-time") == 0) {
			sprintf(tcl.buffer(), "%u", unixtime().tv_sec);
			tcl.result(tcl.buffer());
			return (TCL_OK);
		}
		if (strcmp(argv[1], "media-time") == 0) {
			sprintf(tcl.buffer(), "%u", as_->Clock());
			tcl.result(tcl.buffer());
			return (TCL_OK);
		}
		if (strcmp(argv[1], "active") == 0) {
			tcl.result(active_ ? "1" : "0");
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-input") == 0) {
			sprintf(tcl.buffer(), "%d", as_->AGCLevel() / 10 - 10);
			tcl.result(tcl.buffer());
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-output") == 0) {
			sprintf(tcl.buffer(), "%d", ns_->AGCLevel() / 10 - 10);
			tcl.result(tcl.buffer());
			return (TCL_OK);
		}
	} else if (argc == 3) {
		if (strcmp(argv[1], "audio") == 0) {
			audio_ = (Audio*)TclObject::lookup(argv[2]);
			audio_->attach(this);
			audio_->handler(this);
			update(audio_);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "encoder") == 0) {
			encoder_ = (PCM_Encoder*)TclObject::lookup(argv[2]);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "input-meter") == 0) {
			rmeter_ = (VUMeter*)TclObject::lookup(argv[2]);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "output-meter") == 0) {
			pmeter_ = (VUMeter*)TclObject::lookup(argv[2]);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "silence-thresh") == 0) {
			int thresh = atoi(argv[2]);
			as_->ssthresh(thresh);
			ns_->ssthresh(thresh);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "talk-thresh") == 0) {
			talk_thresh_ = atoi(argv[2]);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "echothresh") == 0) {
			echo_thresh_ = atoi(argv[2]);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "echodelay") == 0) {
			echo_suppress_time_ = atoi(argv[2]) / 20 * FRAMESIZE;
			return (TCL_OK);
		}
		if (strcmp(argv[1], "blocks-per-packet") == 0) {
			outmax_ = atoi(argv[2]) * FRAMESIZE;
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-input") == 0) {
			int level = atoi(argv[2]);
			level = 10 * (level + 10);
			as_->SetAGCLevel(level);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-input-enable") == 0) {
			as_->DoAGC(atoi(argv[2]));
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-output") == 0) {
			int level = atoi(argv[2]);
			level = 10 * (level + 10);
			ns_->SetAGCLevel(level);
			return (TCL_OK);
		}
		if (strcmp(argv[1], "agc-output-enable") == 0) {
			ns_->DoAGC(atoi(argv[2]));
			return (TCL_OK);
		}
		if (strcmp(argv[1], "active") == 0) {
			active_ = atoi(argv[2]);
			return (TCL_OK);
		}
	}
	return (TclObject::command(argc, argv));
}