~ubuntu-branches/debian/sid/terminatorx/sid

/*

    terminatorX - realtime audio scratching software

    Copyright (C) 1999-2011  Alexander König

    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.

    File: tX_aduiodevice.cc

    Description: Implements Audiodevice handling... 

*/    

#define ALSA_PCM_NEW_HW_PARAMS_API

#include "tX_audiodevice.h"

#include "tX_vtt.h"

#include <sys/types.h>

#include <sys/stat.h>

#include <sys/ioctl.h>

#include <fcntl.h>

#include <sys/soundcard.h>

#include <config.h>

#include <string.h>

#include "tX_endian.h"

#ifndef __USE_XOPEN

#	define __USE_XOPEN // we need this for swab()

#	include <unistd.h>

#	undef __USE_XOPEN

#else

#	include <unistd.h>

#endif

#include <string.h>

#include <errno.h>

#include <stdlib.h>

#include "tX_engine.h"

tX_audiodevice :: tX_audiodevice() : samples_per_buffer(0),

current_buffer(0), buffer_pos(0), is_open(false)

{

	sample_buffer[0]=NULL;

	sample_buffer[1]=NULL;

	engine=tX_engine::get_instance();

}

void tX_audiodevice :: start() {

	sample_buffer[0]=new int16_t[samples_per_buffer*2];

	sample_buffer[1]=new int16_t[samples_per_buffer*2];

	int current=0;

	vtt_buffer_size=vtt_class::get_mix_buffer_size()<<1;

	buffer_pos=0;

	while (!engine->is_stopped()) {

		current=current ? 0 : 1;

		int16_t *current_buffer=sample_buffer[current];

		int16_t *next_buffer=sample_buffer[current ? 0 : 1];

		fill_buffer(current_buffer, next_buffer);

		play(current_buffer);

	}

	delete [] sample_buffer[0];

	delete [] sample_buffer[1];

}

void tX_audiodevice :: fill_buffer(int16_t *target_buffer, int16_t *next_target_buffer) {

	int prefill=0;

	while (buffer_pos <= samples_per_buffer) {

		int16_t *data=engine->render_cycle();

		int rest=(buffer_pos+vtt_buffer_size)-samples_per_buffer;

		if (rest<=0) {

			memcpy(&target_buffer[buffer_pos], data, vtt_buffer_size << 1);

		} else {

			memcpy(&target_buffer[buffer_pos], data, (vtt_buffer_size-rest) << 1);

			memcpy(next_target_buffer, &data[vtt_buffer_size-rest], rest << 1);

			prefill=rest;

		}

		buffer_pos+=vtt_buffer_size;

	}

	buffer_pos=prefill;

}

/* Driver Specific Code follows.. */

#ifdef USE_OSS

int tX_audiodevice_oss :: open()

{

	int i=0;

	int p;

	int buff_cfg;

	if (fd) return (1);

	fd=::open(globals.oss_device, O_WRONLY, 0);

	if (fd==-1) {

		tX_error("tX_audiodevice_oss::open() can't open device: %s", strerror(errno));

		return -1;

	}

	is_open=true;

	/* setting buffer size */	

	buff_cfg=(globals.oss_buff_no<<16) | globals.oss_buff_size;

	p=buff_cfg;

	tX_debug("tX_audiodevice_oss::open() - buff_no: %i, buff_size: %i, buff_cfg: %08x", globals.oss_buff_no, globals.oss_buff_size, buff_cfg);

	i = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &p);

	ioctl(fd, SNDCTL_DSP_RESET, 0);		

	/* 16 Bits */

	p =  16;

	i +=  ioctl(fd, SOUND_PCM_WRITE_BITS, &p);

	/* STEREO :) */

	p =  2;

	i += ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &p);

	/* 44.1 khz */

	p =  globals.oss_samplerate;

	i += ioctl(fd, SOUND_PCM_WRITE_RATE, &p);

	sample_rate=globals.oss_samplerate;

	/* Figure actual blocksize.. */

	i += ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &blocksize);

	samples_per_buffer=blocksize/sizeof(int16_t);

	tX_debug("tX_adudiodevice_oss::open() - blocksize: %i, samples_per_buffer: %i", blocksize, samples_per_buffer);

	ioctl(fd, SNDCTL_DSP_SYNC, 0);

	return i;

}

int tX_audiodevice_oss :: close()

{

	if (!fd) {	

		return(1);		

	}

	is_open=false;

	::close(fd);

	fd=0;

	blocksize=0;

	return 0;

}

tX_audiodevice_oss :: tX_audiodevice_oss() : tX_audiodevice(),

fd(0), blocksize(0) {}

void tX_audiodevice_oss :: play(int16_t *buffer)

{

#ifdef BIG_ENDIAN_MACHINE

	swapbuffer (buffer, samples_per_buffer);

#endif

	int res=write(fd, buffer, blocksize);	

	if (res==-1) {

		tX_error("failed to write to audiodevice: %s", strerror(errno));

		exit(-1);

	}

}

#endif //USE_OSS

#ifdef USE_ALSA

#define tX_abs(x) (x<0 ? -x : x)

int tX_audiodevice_alsa :: open()

{

	snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;

	snd_pcm_hw_params_t *hw_params;

	char pcm_name[64];

	char *pos;

	strncpy(pcm_name, globals.alsa_device_id, sizeof(pcm_name));

	if ((pos = strchr(pcm_name, '#')) != NULL) *pos = 0;

	if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0) {

		tX_error("ALSA: Failed to access PCM device \"%s\"", pcm_name);

		return -1;

	}

	is_open=true;

	snd_pcm_hw_params_alloca(&hw_params);	

	if (snd_pcm_hw_params_any(pcm_handle, hw_params) < 0) {

		tX_error("ALSA: Failed to configure PCM device \"%s\"", pcm_name);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	/* Setting INTERLEAVED stereo... */

#ifdef USE_ALSA_MEMCPY

	if (snd_pcm_hw_params_set_access(pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {

#else

	if (snd_pcm_hw_params_set_access(pcm_handle, hw_params, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) {

#endif	

		tX_error("ALSA: Failed to set interleaved access for PCM device \"%s\"", pcm_name);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	/* Make it 16 Bit native endian */

	if (snd_pcm_hw_params_set_format(pcm_handle, hw_params, SND_PCM_FORMAT_S16) < 0) {

		tX_error("ALSA: Error setting 16 Bit sample width for PCM device \"%s\"", pcm_name);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	/* Setting sampling rate */

	unsigned int hw_rate=(unsigned int)globals.alsa_samplerate;

	int dir;

	if (snd_pcm_hw_params_set_rate_near(pcm_handle, hw_params, &hw_rate, &dir) < 0) {

		tX_error("ALSA: Failed setting sample rate: %i", globals.alsa_samplerate);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	if (dir != 0) {

		if (tX_abs(globals.alsa_samplerate - hw_rate) > 2) {

			tX_warning("ALSA: The PCM device \"%s\" doesn\'t support %i Hz playback - using %i instead", pcm_name, globals.alsa_samplerate, hw_rate);

		}

	}	

	sample_rate=hw_rate;

	/* Using stereo output */

	if (snd_pcm_hw_params_set_channels(pcm_handle, hw_params, 2) < 0) {

		tX_error("ALSA: PCM device \"%s\" does not support stereo operation", pcm_name);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	unsigned int buffer_time=globals.alsa_buffer_time;

	unsigned int period_time=globals.alsa_period_time;

	if (snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hw_params, &buffer_time, &dir) < 0) {

		tX_error("ALSA: failed to set the buffer time opf %i usecs", globals.alsa_buffer_time);

		return -1;

	}

	long unsigned int buffer_size;

	if (snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size) < 0) {

		tX_error("ALSA: failed to retreive buffer size");

		return -1;

	}

	tX_debug("ALSA: buffer size is %lu", buffer_size);

	if (snd_pcm_hw_params_set_period_time_near(pcm_handle, hw_params, &period_time, &dir) < 0) {

		tX_error("ALSA: failed to set period time %i", globals.alsa_period_time);

		return -1;

	}

	if (snd_pcm_hw_params_get_period_size(hw_params, &period_size, &dir)<0) {

		tX_error("ALSA: failed to retreive period size");

		return -1;

	}

	samples_per_buffer=period_size;

	/* Apply all that setup work.. */

	if (snd_pcm_hw_params(pcm_handle, hw_params) < 0) {

		tX_error("ALSA: Failed to apply settings to PCM device \"%s\"", pcm_name);

		snd_pcm_hw_params_free (hw_params);

		return -1;

	}

	if (globals.alsa_free_hwstats) {

		snd_pcm_hw_params_free (hw_params);

	}

	return 0; //snd_pcm_prepare(pcm_handle);

}

int tX_audiodevice_alsa :: close()

{

	if (is_open) {

		snd_pcm_close(pcm_handle);

	}

	is_open=false;

	return 0;

}

tX_audiodevice_alsa :: tX_audiodevice_alsa() : tX_audiodevice(),

pcm_handle(NULL) {}

void tX_audiodevice_alsa :: play(int16_t *buffer)

{

	int underrun_ctr=0;

	snd_pcm_sframes_t pcmreturn;

#ifdef USE_ALSA_MEMCPY

	pcmreturn = snd_pcm_writei(pcm_handle, buffer, samples_per_buffer >> 1);

#else	

	pcmreturn = snd_pcm_mmap_writei(pcm_handle, buffer, samples_per_buffer >> 1);

#endif

	while (pcmreturn==-EPIPE) {

		snd_pcm_prepare(pcm_handle);

#ifdef USE_ALSA_MEMCPY

		pcmreturn = snd_pcm_writei(pcm_handle, buffer, samples_per_buffer >> 1);

#else	

		pcmreturn = snd_pcm_mmap_writei(pcm_handle, buffer, samples_per_buffer >> 1);

#endif

		underrun_ctr++;

		if (underrun_ctr>100) {

			tX_error("tX_audiodevice_alsa::play() more than 10 EPIPE cycles. Giving up.");

			break;

		}

		//tX_warning("ALSA: ** buffer underrun **");

	}

	if (pcmreturn<0) {

		printf("snd_pcm_writei says: %s.\n", strerror(-1*pcmreturn));

	}

}

#endif //USE_ALSA

#ifdef USE_JACK

tX_jack_client* tX_jack_client::instance=NULL;

void tX_jack_client::init()

{

		tX_jack_client *test=new tX_jack_client();

		if (!instance) {

			delete test;

		}	

}

tX_jack_client::tX_jack_client():device(NULL),jack_shutdown(false)

{

	jack_set_error_function(tX_jack_client::error);

	if ((client=jack_client_open("terminatorX", (jack_options_t) NULL, NULL))==0) {

		tX_error("tX_jack_client() -> failed to connect to jackd.");

		instance=NULL;

	} else {

		instance=this;

		const char **ports;

		/* Setting up jack callbacks... */		

		jack_set_process_callback(client, tX_jack_client::process, NULL);

		jack_set_sample_rate_callback(client, tX_jack_client::srate, NULL);		

		jack_on_shutdown (client, tX_jack_client::shutdown, NULL);

		/* Creating the port... */

		left_port = jack_port_register (client, "output_1", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);

		right_port = jack_port_register (client, "output_2", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);

		/* Action... */

		jack_activate(client);

		/* Connect some ports... */

		if ((ports = jack_get_ports (client, NULL, NULL, JackPortIsPhysical|JackPortIsInput)) == NULL) {

			tX_error("tX_jack_client() no ports to connect to found. Connect manually.");

		} else if (ports[0] && ports[1]) {

			if (jack_connect (client, jack_port_name(left_port), ports[0])) {

				tX_error("tX_jack_client() failed to connect left port.");

			}

			if (jack_connect (client, jack_port_name(right_port), ports[1])) {

				tX_error("tX_jack_client() failed to connect right port.");

			}

			free (ports);

		}

	}

}

tX_jack_client::~tX_jack_client()

{

	instance=NULL;

	if (client) jack_client_close(client);

}

void tX_jack_client::error(const char *desc)

{

	tX_error("jack error: %s.", desc);

}

int tX_jack_client::srate(jack_nframes_t nframes, void *arg)

{

	tX_error("tX_jack_client::srate() jack changed samplerate - ignored.");

	return 0;

}

void tX_jack_client::shutdown(void *arg)

{

	tX_error("tX_jack_client::shutdown() jack daemon has shut down. Bad!");

	if (instance) instance->jack_shutdown=true;

}

int tX_jack_client::process(jack_nframes_t nframes, void *arg)

{

	if (instance) {

		return instance->play(nframes);

	}

	/* Hmm, what to do in such a case? */

	return 0;

}

int tX_jack_client::play(jack_nframes_t nframes)

{

	jack_default_audio_sample_t *left = (jack_default_audio_sample_t *) jack_port_get_buffer (left_port, nframes);

	jack_default_audio_sample_t *right = (jack_default_audio_sample_t *) jack_port_get_buffer (right_port, nframes);

	if (device) {

		device->fill_frames(left, right, nframes);

	} else {

		memset(left, 0, sizeof (jack_default_audio_sample_t) * nframes);

		memset(right, 0, sizeof (jack_default_audio_sample_t) * nframes);

	}

	return 0;

}

int tX_jack_client::get_sample_rate() 

{

	return jack_get_sample_rate(client);

}

/* tX_audiodevice_jack */

tX_audiodevice_jack::tX_audiodevice_jack():tX_audiodevice()

{

	client=NULL;

}

int tX_audiodevice_jack::open()

{

	tX_jack_client *jack_client=tX_jack_client::get_instance();

	if (jack_client) {

		sample_rate=jack_client->get_sample_rate();

		client=jack_client;

		is_open=true;

		return 0;

	}

	return 1;

}

int tX_audiodevice_jack::close()

{

	if (client) {

		client->set_device(NULL);

	}

	is_open=false;	

	return 0;

}

void tX_audiodevice_jack::play(int16_t *buffer)

{

	tX_error("tX_audiodevice_jack::play()");

}

void tX_audiodevice_jack::start()

{

	overrun_buffer=new f_prec[vtt_class::samples_in_mix_buffer];

	client->set_device(this);

	while ((!engine->is_stopped()) && !(client->get_jack_shutdown())) {

		usleep(100);

	}	

	client->set_device(NULL);

	delete [] overrun_buffer;

}

void tX_audiodevice_jack::fill_frames(jack_default_audio_sample_t *left, jack_default_audio_sample_t *right, jack_nframes_t nframes)

{

	unsigned int outbuffer_pos=0;

	unsigned int sample;

	if (samples_in_overrun_buffer) {

		for (sample=0; ((sample<samples_in_overrun_buffer) && (outbuffer_pos<nframes));) {

			left[outbuffer_pos]=overrun_buffer[sample++]/32767.0;

			right[outbuffer_pos++]=overrun_buffer[sample++]/32767.0;

		}

	}

	while (outbuffer_pos<nframes) {

		engine->render_cycle();

		for (sample=0; ((sample<(unsigned int) vtt_class::samples_in_mix_buffer) && (outbuffer_pos<nframes));) {

			left[outbuffer_pos]=vtt_class::mix_buffer[sample++]/32767.0;

			right[outbuffer_pos++]=vtt_class::mix_buffer[sample++]/32767.0;

		}

		if (sample<(unsigned int) vtt_class::samples_in_mix_buffer) {

			samples_in_overrun_buffer=vtt_class::samples_in_mix_buffer-sample;

			/* There's more data in the mixbuffer... */

			memcpy(overrun_buffer, &vtt_class::mix_buffer[sample], sizeof(f_prec) * samples_in_overrun_buffer);

		} else {

			samples_in_overrun_buffer=0;

		}

	}

}

#endif // USE_JACK