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1724 lines
53 KiB
1724 lines
53 KiB
/*
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ALSA player.
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-------------------
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Copyright:
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(C) 2005 by Gary Cramblitt <garycramblitt@comcast.net>
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Portions based on aplay.c in alsa-utils
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Copyright (c) by Jaroslav Kysela <perex@suse.cz>
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Based on vplay program by Michael Beck
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-------------------
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Original author: Gary Cramblitt <garycramblitt@comcast.net>
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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
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along with this program; if not, write to the Free Software
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Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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******************************************************************************/
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// AlsaPlayer includes.
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#include "alsaplayer.h"
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// #include <sys/wait.h>
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// System includes.
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#if defined(HAVE_SYS_TIME_H)
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# include <sys/time.h>
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#endif
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#if defined(TIME_WITH_SYS_TIME)
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# include <time.h>
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#endif
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// TQt includes.
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#include <tqdir.h>
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#include <tqapplication.h>
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#include <tqcstring.h>
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// KDE includes.
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#include <kdebug.h>
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#include <tdeconfig.h>
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#include <kstandarddirs.h>
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#include <tdemessagebox.h>
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#include <tdelocale.h>
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#if !defined(__GNUC__) || __GNUC__ >= 3
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#define ERR(...) do {\
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TQString dbgStr;\
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TQString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( __VA_ARGS__); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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} while (0)
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#else
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#define ERR(args...) do {\
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TQString dbgStr;\
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TQString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( ##args ); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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} while (0)
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#endif
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#if !defined(__GNUC__) || __GNUC__ >= 3
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#define MSG(...) do {\
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if (m_debugLevel >= 1) {\
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TQString dbgStr; \
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TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( __VA_ARGS__); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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}; \
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} while (0)
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#else
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#define MSG(args...) do {\
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if (m_debugLevel >= 1) {\
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TQString dbgStr; \
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TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( ##args ); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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}; \
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} while (0)
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#endif
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#if !defined(__GNUC__) || __GNUC__ >= 3
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#define DBG(...) do {\
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if (m_debugLevel >= 2) {\
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TQString dbgStr; \
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TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( __VA_ARGS__); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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}; \
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} while (0)
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#else
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#define DBG(args...) do {\
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if (m_debugLevel >= 2) {\
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TQString dbgStr; \
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TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \
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s += dbgStr.sprintf( ##args ); \
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kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \
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}; \
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} while (0)
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#endif
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TQString AlsaPlayer::timestamp() const
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{
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time_t t;
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struct timeval tv;
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char *tstr;
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t = time(NULL);
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tstr = strdup(ctime(&t));
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tstr[strlen(tstr)-1] = 0;
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gettimeofday(&tv,NULL);
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TQString ts;
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ts.sprintf(" %s [%d] ",tstr, (int) tv.tv_usec);
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free(tstr);
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return ts;
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}
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////////////////////////////////////////////////////////////////////////////////
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// public methods
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////////////////////////////////////////////////////////////////////////////////
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AlsaPlayer::AlsaPlayer(TQObject* parent, const char* name, const TQStringList& args) :
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Player(parent, name, args),
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m_currentVolume(1.0),
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m_pcmName("default"),
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m_defPeriodSize(128),
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m_defPeriods(8),
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m_debugLevel(1),
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m_simulatedPause(false)
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{
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init();
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}
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AlsaPlayer::~AlsaPlayer()
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{
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if (running()) {
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stop();
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wait();
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}
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}
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//void AlsaPlayer::play(const FileHandle &file)
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void AlsaPlayer::startPlay(const TQString &file)
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{
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if (running()) {
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if (paused()) {
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if (canPause)
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snd_pcm_pause(handle, false);
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else
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m_simulatedPause = false;
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}
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return;
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}
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audiofile.setName(file);
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audiofile.open(IO_ReadOnly);
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fd = audiofile.handle();
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// Start thread running.
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start();
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}
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/*virtual*/ void AlsaPlayer::run()
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{
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TQString pName = m_pcmName.section(" ", 0, 0);
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DBG("pName = %s", pName.ascii());
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pcm_name = tqstrdup(pName.ascii());
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int err;
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snd_pcm_info_t *info;
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m_simulatedPause = false;
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snd_pcm_info_alloca(&info);
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err = snd_output_stdio_attach(&log, stderr, 0);
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assert(err >= 0);
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rhwdata.format = DEFAULT_FORMAT;
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rhwdata.rate = DEFAULT_SPEED;
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rhwdata.channels = 1;
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err = snd_pcm_open(&handle, pcm_name, stream, open_mode);
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if (err < 0) {
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ERR("audio open error on pcm device %s: %s", pcm_name, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_info(handle, info)) < 0) {
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ERR("info error: %s", snd_strerror(err));
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return;
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}
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chunk_size = 1024;
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hwdata = rhwdata;
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audioBuffer.resize(1024);
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// audiobuf = (char *)malloc(1024);
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audiobuf = audioBuffer.data();
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if (audiobuf == NULL) {
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ERR("not enough memory");
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return;
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}
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if (mmap_flag) {
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writei_func = snd_pcm_mmap_writei;
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readi_func = snd_pcm_mmap_readi;
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writen_func = snd_pcm_mmap_writen;
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readn_func = snd_pcm_mmap_readn;
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} else {
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writei_func = snd_pcm_writei;
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readi_func = snd_pcm_readi;
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writen_func = snd_pcm_writen;
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readn_func = snd_pcm_readn;
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}
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playback(fd);
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cleanup();
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return;
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}
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void AlsaPlayer::pause()
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{
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if (running()) {
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DBG("Pause requested");
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m_mutex.lock();
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if (handle) {
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// Some hardware can pause; some can't. canPause is set in set_params.
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if (canPause) {
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m_simulatedPause = false;
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snd_pcm_pause(handle, true);
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m_mutex.unlock();
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} else {
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// Set a flag and cause wait_for_poll to sleep. When resumed, will get
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// an underrun.
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m_simulatedPause = true;
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m_mutex.unlock();
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}
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}
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}
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}
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void AlsaPlayer::stop()
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{
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if (running()) {
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DBG("STOP! Locking mutex");
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m_mutex.lock();
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m_simulatedPause = false;
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if (handle) {
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/* This constant is arbitrary */
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char buf = 42;
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DBG("Request for stop, device state is %s",
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snd_pcm_state_name(snd_pcm_state(handle)));
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write(alsa_stop_pipe[1], &buf, 1);
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}
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DBG("unlocking mutex");
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m_mutex.unlock();
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/* Wait for thread to exit */
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DBG("waiting for thread to exit");
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wait();
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DBG("cleaning up");
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}
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cleanup();
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}
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/*
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* Stop playback, cleanup and exit thread.
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*/
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void AlsaPlayer::stopAndExit()
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{
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// if (handle) snd_pcm_drop(handle);
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cleanup();
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exit();
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}
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void AlsaPlayer::setVolume(float volume)
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{
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m_currentVolume = volume;
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}
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float AlsaPlayer::volume() const
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{
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return m_currentVolume;
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}
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/////////////////////////////////////////////////////////////////////////////////
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// player status functions
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/////////////////////////////////////////////////////////////////////////////////
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bool AlsaPlayer::playing() const
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{
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bool result = false;
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if (running()) {
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m_mutex.lock();
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if (handle) {
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if (canPause) {
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snd_pcm_status_t *status;
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snd_pcm_status_alloca(&status);
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int res;
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if ((res = snd_pcm_status(handle, status)) < 0)
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ERR("status error: %s", snd_strerror(res));
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else {
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result = (SND_PCM_STATE_RUNNING == snd_pcm_status_get_state(status))
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|| (SND_PCM_STATE_DRAINING == snd_pcm_status_get_state(status));
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DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
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}
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} else
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result = !m_simulatedPause;
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}
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m_mutex.unlock();
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}
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return result;
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}
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bool AlsaPlayer::paused() const
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{
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bool result = false;
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if (running()) {
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m_mutex.lock();
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if (handle) {
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if (canPause) {
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snd_pcm_status_t *status;
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snd_pcm_status_alloca(&status);
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int res;
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if ((res = snd_pcm_status(handle, status)) < 0)
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ERR("status error: %s", snd_strerror(res));
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else {
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result = (SND_PCM_STATE_PAUSED == snd_pcm_status_get_state(status));
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DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
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}
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} else
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result = m_simulatedPause;
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}
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m_mutex.unlock();
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}
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return result;
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}
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int AlsaPlayer::totalTime() const
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{
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int total = 0;
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int rate = hwdata.rate;
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int channels = hwdata.channels;
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if (rate > 0 && channels > 0) {
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total = int((double(pbrec_count) / rate) / channels);
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// DBG("pbrec_count = %i rate =%i channels = %i", pbrec_count, rate, channels);
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// DBG("totalTime = %i", total);
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}
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return total;
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}
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int AlsaPlayer::currentTime() const
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{
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int current = 0;
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int rate = hwdata.rate;
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int channels = hwdata.channels;
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if (rate > 0 && channels > 0) {
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current = int((double(fdcount) / rate) / channels);
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// DBG("fdcount = %i rate = %i channels = %i", fdcount, rate, channels);
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// DBG("currentTime = %i", current);
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}
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return current;
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}
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int AlsaPlayer::position() const
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{
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// TODO: Make this more accurate by adding frames that have been so-far
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// played within the Alsa ring buffer.
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return pbrec_count > 0 ? int(double(fdcount) * 1000 / pbrec_count + .5) : 0;
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}
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/////////////////////////////////////////////////////////////////////////////////
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// player seek functions
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/////////////////////////////////////////////////////////////////////////////////
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void AlsaPlayer::seek(int /*seekTime*/)
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{
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// TODO:
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}
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void AlsaPlayer::seekPosition(int /*position*/)
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{
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// TODO:
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}
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/*
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* Returns a list of PCM devices.
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* This function fills the specified list with ALSA hardware soundcards found on the system.
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* It uses plughw:xx instead of hw:xx for specifiers, because hw:xx are not practical to
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* use (e.g. they require a resampler/channel mixer in the application).
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*/
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TQStringList AlsaPlayer::getPluginList( const TQCString& /*classname*/ )
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{
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int err = 0;
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int card = -1, device = -1;
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snd_ctl_t *handle;
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snd_ctl_card_info_t *info;
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snd_pcm_info_t *pcminfo;
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snd_ctl_card_info_alloca(&info);
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snd_pcm_info_alloca(&pcminfo);
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TQStringList result;
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result.append("default");
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for (;;) {
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err = snd_card_next(&card);
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if (err < 0 || card < 0) break;
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if (card >= 0) {
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char name[32];
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sprintf(name, "hw:%i", card);
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if ((err = snd_ctl_open(&handle, name, 0)) < 0) continue;
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if ((err = snd_ctl_card_info(handle, info)) < 0) {
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snd_ctl_close(handle);
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continue;
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}
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for (int devCnt=0;;++devCnt) {
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err = snd_ctl_pcm_next_device(handle, &device);
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if (err < 0 || device < 0) break;
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snd_pcm_info_set_device(pcminfo, device);
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snd_pcm_info_set_subdevice(pcminfo, 0);
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snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_PLAYBACK);
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if ((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) continue;
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TQString infoName = " ";
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infoName += snd_ctl_card_info_get_name(info);
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infoName += " (";
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infoName += snd_pcm_info_get_name(pcminfo);
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infoName += ")";
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if (0 == devCnt) {
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TQString pcmName = TQString("default:%1").arg(card);
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result.append(pcmName + infoName);
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}
|
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TQString pcmName = TQString("plughw:%1,%2").arg(card).arg(device);
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result.append(pcmName + infoName);
|
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}
|
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snd_ctl_close(handle);
|
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}
|
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}
|
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return result;
|
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}
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|
|
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// TQStringList AlsaPlayer::getPluginList( const TQCString& /*classname*/ )
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// {
|
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// TQStringList assumed("default");
|
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// snd_config_t *conf;
|
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// int err = snd_config_update();
|
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// if (err < 0) {
|
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// ERR("snd_config_update: %s", snd_strerror(err));
|
|
// return assumed;
|
|
// }
|
|
// err = snd_config_search(snd_config, "pcm", &conf);
|
|
// if (err < 0) return TQStringList();
|
|
// snd_config_iterator_t it = snd_config_iterator_first(conf);
|
|
// snd_config_iterator_t itEnd = snd_config_iterator_end(conf);
|
|
// const char* id;
|
|
// snd_config_t *entry;
|
|
// TQStringList result;
|
|
// snd_ctl_card_info_t *info;
|
|
// snd_ctl_card_info_alloca(&info);
|
|
// snd_pcm_info_t *pcminfo;
|
|
// snd_pcm_info_alloca(&pcminfo);
|
|
// while (it != itEnd) {
|
|
// entry = snd_config_iterator_entry(it);
|
|
// err = snd_config_get_id(entry, &id);
|
|
// if (err >= 0) {
|
|
// if (TQString(id) != "default")
|
|
// {
|
|
// int card = -1;
|
|
// while (snd_card_next(&card) >= 0 && card >= 0) {
|
|
// char name[32];
|
|
// sprintf(name, "%s:%d", id, card);
|
|
// DBG("Checking %s", name);
|
|
// snd_ctl_t *handle;
|
|
// if ((err = snd_ctl_open(&handle, name, SND_CTL_NONBLOCK)) >= 0) {
|
|
// if ((err = snd_ctl_card_info(handle, info)) >= 0) {
|
|
// int dev = -1;
|
|
// snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
|
|
// while (snd_ctl_pcm_next_device(handle, &dev) >= 0 && dev >= 0) {
|
|
// snd_pcm_info_set_device(pcminfo, dev);
|
|
// snd_pcm_info_set_subdevice(pcminfo, 0);
|
|
// snd_pcm_info_set_stream(pcminfo, stream);
|
|
// if ((err = snd_ctl_pcm_info(handle, pcminfo)) >= 0) {
|
|
// TQString pluginName = name;
|
|
// pluginName += ",";
|
|
// pluginName += TQString::number(dev);
|
|
// pluginName += " ";
|
|
// pluginName += snd_ctl_card_info_get_name(info);
|
|
// pluginName += ",";
|
|
// pluginName += snd_pcm_info_get_name(pcminfo);
|
|
// result.append(pluginName);
|
|
// // DBG(pluginName);
|
|
// }
|
|
// }
|
|
// }
|
|
// snd_ctl_close(handle);
|
|
// }
|
|
// }
|
|
// if (card == -1) result.append(id);
|
|
// } else result.append(id);
|
|
// }
|
|
// it = snd_config_iterator_next(it);
|
|
// }
|
|
// snd_config_update_free_global();
|
|
// return result;
|
|
// }
|
|
|
|
void AlsaPlayer::setSinkName(const TQString& sinkName) { m_pcmName = sinkName; }
|
|
|
|
/////////////////////////////////////////////////////////////////////////////////
|
|
// private
|
|
/////////////////////////////////////////////////////////////////////////////////
|
|
|
|
void AlsaPlayer::init()
|
|
{
|
|
pcm_name = 0;
|
|
handle = 0;
|
|
canPause = false;
|
|
timelimit = 0;
|
|
file_type = FORMAT_DEFAULT;
|
|
sleep_min = 0;
|
|
// open_mode = 0;
|
|
open_mode = SND_PCM_NONBLOCK;
|
|
stream = SND_PCM_STREAM_PLAYBACK;
|
|
mmap_flag = 0;
|
|
interleaved = 1;
|
|
audiobuf = NULL;
|
|
chunk_size = 0;
|
|
period_time = 0;
|
|
buffer_time = 0;
|
|
avail_min = -1;
|
|
start_delay = 0;
|
|
stop_delay = 0;
|
|
buffer_pos = 0;
|
|
log = 0;
|
|
fd = -1;
|
|
pbrec_count = LLONG_MAX;
|
|
alsa_stop_pipe[0] = 0;
|
|
alsa_stop_pipe[1] = 0;
|
|
alsa_poll_fds = 0;
|
|
m_simulatedPause = false;
|
|
}
|
|
|
|
void AlsaPlayer::cleanup()
|
|
{
|
|
DBG("cleaning up");
|
|
m_mutex.lock();
|
|
if (pcm_name) free(pcm_name);
|
|
if (fd >= 0) audiofile.close();
|
|
if (handle) {
|
|
snd_pcm_drop(handle);
|
|
snd_pcm_close(handle);
|
|
}
|
|
if (alsa_stop_pipe[0]) close(alsa_stop_pipe[0]);
|
|
if (alsa_stop_pipe[1]) close(alsa_stop_pipe[1]);
|
|
if (audiobuf) audioBuffer.resize(0);
|
|
if (alsa_poll_fds) alsa_poll_fds_barray.resize(0);
|
|
if (log) snd_output_close(log);
|
|
snd_config_update_free_global();
|
|
init();
|
|
m_mutex.unlock();
|
|
}
|
|
|
|
/*
|
|
* Safe read (for pipes)
|
|
*/
|
|
|
|
ssize_t AlsaPlayer::safe_read(int fd, void *buf, size_t count)
|
|
{
|
|
ssize_t result = 0;
|
|
ssize_t res;
|
|
|
|
while (count > 0) {
|
|
if ((res = read(fd, buf, count)) == 0)
|
|
break;
|
|
if (res < 0)
|
|
return result > 0 ? result : res;
|
|
count -= res;
|
|
result += res;
|
|
buf = (char *)buf + res;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/*
|
|
* Test, if it is a .VOC file and return >=0 if ok (this is the length of rest)
|
|
* < 0 if not
|
|
*/
|
|
int AlsaPlayer::test_vocfile(void *buffer)
|
|
{
|
|
VocHeader *vp = (VocHeader*)buffer;
|
|
|
|
if (!memcmp(vp->magic, VOC_MAGIC_STRING, 20)) {
|
|
vocminor = LE_SHORT(vp->version) & 0xFF;
|
|
vocmajor = LE_SHORT(vp->version) / 256;
|
|
if (LE_SHORT(vp->version) != (0x1233 - LE_SHORT(vp->coded_ver)))
|
|
return -2; /* coded version mismatch */
|
|
return LE_SHORT(vp->headerlen) - sizeof(VocHeader); /* 0 mostly */
|
|
}
|
|
return -1; /* magic string fail */
|
|
}
|
|
|
|
/*
|
|
* helper for test_wavefile
|
|
*/
|
|
|
|
size_t AlsaPlayer::test_wavefile_read(int fd, char *buffer, size_t *size, size_t reqsize, int line)
|
|
{
|
|
if (*size >= reqsize)
|
|
return *size;
|
|
if ((size_t)safe_read(fd, buffer + *size, reqsize - *size) != reqsize - *size) {
|
|
ERR("read error (called from line %i)", line);
|
|
stopAndExit();
|
|
}
|
|
return *size = reqsize;
|
|
}
|
|
|
|
#define check_wavefile_space(buffer, len, blimit) \
|
|
if (len > blimit) { \
|
|
blimit = len; \
|
|
if ((buffer = (char*)realloc(buffer, blimit)) == NULL) { \
|
|
ERR("not enough memory"); \
|
|
stopAndExit(); \
|
|
} \
|
|
}
|
|
|
|
/*
|
|
* test, if it's a .WAV file, > 0 if ok (and set the speed, stereo etc.)
|
|
* == 0 if not
|
|
* Value returned is bytes to be discarded.
|
|
*/
|
|
ssize_t AlsaPlayer::test_wavefile(int fd, char *_buffer, size_t size)
|
|
{
|
|
WaveHeader *h = (WaveHeader *)_buffer;
|
|
char *buffer = NULL;
|
|
size_t blimit = 0;
|
|
WaveFmtBody *f;
|
|
WaveChunkHeader *c;
|
|
u_int type;
|
|
u_int len;
|
|
|
|
if (size < sizeof(WaveHeader))
|
|
return -1;
|
|
if (h->magic != WAV_RIFF || h->type != WAV_WAVE)
|
|
return -1;
|
|
if (size > sizeof(WaveHeader)) {
|
|
check_wavefile_space(buffer, size - sizeof(WaveHeader), blimit);
|
|
memcpy(buffer, _buffer + sizeof(WaveHeader), size - sizeof(WaveHeader));
|
|
}
|
|
size -= sizeof(WaveHeader);
|
|
while (1) {
|
|
check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
|
|
test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
|
|
c = (WaveChunkHeader*)buffer;
|
|
type = c->type;
|
|
len = LE_INT(c->length);
|
|
len += len % 2;
|
|
if (size > sizeof(WaveChunkHeader))
|
|
memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
|
|
size -= sizeof(WaveChunkHeader);
|
|
if (type == WAV_FMT)
|
|
break;
|
|
check_wavefile_space(buffer, len, blimit);
|
|
test_wavefile_read(fd, buffer, &size, len, __LINE__);
|
|
if (size > len)
|
|
memmove(buffer, buffer + len, size - len);
|
|
size -= len;
|
|
}
|
|
|
|
if (len < sizeof(WaveFmtBody)) {
|
|
ERR("unknown length of 'fmt ' chunk (read %u, should be %u at least)", len, (u_int)sizeof(WaveFmtBody));
|
|
stopAndExit();
|
|
}
|
|
check_wavefile_space(buffer, len, blimit);
|
|
test_wavefile_read(fd, buffer, &size, len, __LINE__);
|
|
f = (WaveFmtBody*) buffer;
|
|
if (LE_SHORT(f->format) != WAV_PCM_CODE) {
|
|
ERR("can't play not PCM-coded WAVE-files");
|
|
stopAndExit();
|
|
}
|
|
if (LE_SHORT(f->modus) < 1) {
|
|
ERR("can't play WAVE-files with %d tracks", LE_SHORT(f->modus));
|
|
stopAndExit();
|
|
}
|
|
hwdata.channels = LE_SHORT(f->modus);
|
|
switch (LE_SHORT(f->bit_p_spl)) {
|
|
case 8:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_U8)
|
|
MSG("Warning: format is changed to U8");
|
|
hwdata.format = SND_PCM_FORMAT_U8;
|
|
break;
|
|
case 16:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_S16_LE)
|
|
MSG("Warning: format is changed to S16_LE");
|
|
hwdata.format = SND_PCM_FORMAT_S16_LE;
|
|
break;
|
|
case 24:
|
|
switch (LE_SHORT(f->byte_p_spl) / hwdata.channels) {
|
|
case 3:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_S24_3LE)
|
|
MSG("Warning: format is changed to S24_3LE");
|
|
hwdata.format = SND_PCM_FORMAT_S24_3LE;
|
|
break;
|
|
case 4:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_S24_LE)
|
|
MSG("Warning: format is changed to S24_LE");
|
|
hwdata.format = SND_PCM_FORMAT_S24_LE;
|
|
break;
|
|
default:
|
|
ERR("can't play WAVE-files with sample %d bits in %d bytes wide (%d channels)", LE_SHORT(f->bit_p_spl), LE_SHORT(f->byte_p_spl), hwdata.channels);
|
|
stopAndExit();
|
|
}
|
|
break;
|
|
case 32:
|
|
hwdata.format = SND_PCM_FORMAT_S32_LE;
|
|
break;
|
|
default:
|
|
ERR("can't play WAVE-files with sample %d bits wide", LE_SHORT(f->bit_p_spl));
|
|
stopAndExit();
|
|
}
|
|
hwdata.rate = LE_INT(f->sample_fq);
|
|
|
|
if (size > len)
|
|
memmove(buffer, buffer + len, size - len);
|
|
size -= len;
|
|
|
|
while (1) {
|
|
u_int type, len;
|
|
|
|
check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
|
|
test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
|
|
c = (WaveChunkHeader*)buffer;
|
|
type = c->type;
|
|
len = LE_INT(c->length);
|
|
if (size > sizeof(WaveChunkHeader))
|
|
memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
|
|
size -= sizeof(WaveChunkHeader);
|
|
if (type == WAV_DATA) {
|
|
if (len < pbrec_count && len < 0x7ffffffe)
|
|
pbrec_count = len;
|
|
if (size > 0)
|
|
memcpy(_buffer, buffer, size);
|
|
free(buffer);
|
|
return size;
|
|
}
|
|
len += len % 2;
|
|
check_wavefile_space(buffer, len, blimit);
|
|
test_wavefile_read(fd, buffer, &size, len, __LINE__);
|
|
if (size > len)
|
|
memmove(buffer, buffer + len, size - len);
|
|
size -= len;
|
|
}
|
|
|
|
/* shouldn't be reached */
|
|
return -1;
|
|
}
|
|
|
|
/*
|
|
* Test for AU file.
|
|
*/
|
|
|
|
int AlsaPlayer::test_au(int fd, char *buffer)
|
|
{
|
|
AuHeader *ap = (AuHeader*)buffer;
|
|
|
|
if (ap->magic != AU_MAGIC)
|
|
return -1;
|
|
if (BE_INT(ap->hdr_size) > 128 || BE_INT(ap->hdr_size) < 24)
|
|
return -1;
|
|
pbrec_count = BE_INT(ap->data_size);
|
|
switch (BE_INT(ap->encoding)) {
|
|
case AU_FMT_ULAW:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_MU_LAW)
|
|
MSG("Warning: format is changed to MU_LAW");
|
|
hwdata.format = SND_PCM_FORMAT_MU_LAW;
|
|
break;
|
|
case AU_FMT_LIN8:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_U8)
|
|
MSG("Warning: format is changed to U8");
|
|
hwdata.format = SND_PCM_FORMAT_U8;
|
|
break;
|
|
case AU_FMT_LIN16:
|
|
if (hwdata.format != DEFAULT_FORMAT &&
|
|
hwdata.format != SND_PCM_FORMAT_S16_BE)
|
|
MSG("Warning: format is changed to S16_BE");
|
|
hwdata.format = SND_PCM_FORMAT_S16_BE;
|
|
break;
|
|
default:
|
|
return -1;
|
|
}
|
|
hwdata.rate = BE_INT(ap->sample_rate);
|
|
if (hwdata.rate < 2000 || hwdata.rate > 256000)
|
|
return -1;
|
|
hwdata.channels = BE_INT(ap->channels);
|
|
if (hwdata.channels < 1 || hwdata.channels > 128)
|
|
return -1;
|
|
if ((size_t)safe_read(fd, buffer + sizeof(AuHeader), BE_INT(ap->hdr_size) - sizeof(AuHeader)) != BE_INT(ap->hdr_size) - sizeof(AuHeader)) {
|
|
ERR("read error");
|
|
stopAndExit();
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void AlsaPlayer::set_params(void)
|
|
{
|
|
snd_pcm_hw_params_t *hwparams;
|
|
snd_pcm_uframes_t period_size;
|
|
int err;
|
|
int dir;
|
|
unsigned int rate;
|
|
unsigned int periods;
|
|
|
|
snd_pcm_hw_params_alloca(&hwparams);
|
|
err = snd_pcm_hw_params_any(handle, hwparams);
|
|
if (err < 0) {
|
|
ERR("Broken configuration for this PCM: no configurations available");
|
|
stopAndExit();
|
|
}
|
|
|
|
/* Create the pipe for communication about stop requests. */
|
|
if (pipe(alsa_stop_pipe)) {
|
|
ERR("Stop pipe creation failed (%s)", strerror(errno));
|
|
stopAndExit();
|
|
}
|
|
|
|
/* Find how many descriptors we will get for poll(). */
|
|
alsa_fd_count = snd_pcm_poll_descriptors_count(handle);
|
|
if (alsa_fd_count <= 0){
|
|
ERR("Invalid poll descriptors count returned from ALSA.");
|
|
stopAndExit();
|
|
}
|
|
|
|
/* Create and fill in struct pollfd *alsa_poll_fds with ALSA descriptors. */
|
|
// alsa_poll_fds = (pollfd *)malloc ((alsa_fd_count + 1) * sizeof(struct pollfd));
|
|
alsa_poll_fds_barray.resize((alsa_fd_count + 1) * sizeof(struct pollfd));
|
|
alsa_poll_fds = (pollfd *)alsa_poll_fds_barray.data();
|
|
assert(alsa_poll_fds);
|
|
if ((err = snd_pcm_poll_descriptors(handle, alsa_poll_fds, alsa_fd_count)) < 0) {
|
|
ERR("Unable to obtain poll descriptors for playback: %s", snd_strerror(err));
|
|
stopAndExit();
|
|
}
|
|
|
|
/* Create a new pollfd structure for requests by alsa_stop(). */
|
|
struct pollfd alsa_stop_pipe_pfd;
|
|
alsa_stop_pipe_pfd.fd = alsa_stop_pipe[0];
|
|
alsa_stop_pipe_pfd.events = POLLIN;
|
|
alsa_stop_pipe_pfd.revents = 0;
|
|
|
|
/* Join this our own pollfd to the ALSAs ones. */
|
|
alsa_poll_fds[alsa_fd_count] = alsa_stop_pipe_pfd;
|
|
++alsa_fd_count;
|
|
|
|
if (mmap_flag) {
|
|
snd_pcm_access_mask_t *mask = (snd_pcm_access_mask_t *)alloca(snd_pcm_access_mask_sizeof());
|
|
snd_pcm_access_mask_none(mask);
|
|
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
|
|
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
|
|
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
|
|
err = snd_pcm_hw_params_set_access_mask(handle, hwparams, mask);
|
|
} else if (interleaved)
|
|
err = snd_pcm_hw_params_set_access(handle, hwparams,
|
|
SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
else
|
|
err = snd_pcm_hw_params_set_access(handle, hwparams,
|
|
SND_PCM_ACCESS_RW_NONINTERLEAVED);
|
|
if (err < 0) {
|
|
ERR("Error setting access type: %s", snd_strerror(err));
|
|
stopAndExit();
|
|
}
|
|
err = snd_pcm_hw_params_set_format(handle, hwparams, hwdata.format);
|
|
if (err < 0) {
|
|
ERR("Error setting sample format to %i: %s", hwdata.format, snd_strerror(err));
|
|
stopAndExit();
|
|
}
|
|
err = snd_pcm_hw_params_set_channels(handle, hwparams, hwdata.channels);
|
|
if (err < 0) {
|
|
ERR("Error setting channel count to %i: %s", hwdata.channels, snd_strerror(err));
|
|
stopAndExit();
|
|
}
|
|
|
|
#if 0
|
|
err = snd_pcm_hw_params_set_periods_min(handle, hwparams, 2);
|
|
assert(err >= 0);
|
|
#endif
|
|
rate = hwdata.rate;
|
|
#if SND_LIB_MAJOR >= 1
|
|
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &hwdata.rate, 0);
|
|
#else
|
|
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, hwdata.rate, 0);
|
|
#endif
|
|
assert(err >= 0);
|
|
if ((float)rate * 1.05 < hwdata.rate || (float)rate * 0.95 > hwdata.rate) {
|
|
MSG("Warning: rate is not accurate (requested = %iHz, got = %iHz)", rate, hwdata.rate);
|
|
MSG(" please, try the plug plugin (-Dplug:%s)", snd_pcm_name(handle));
|
|
}
|
|
|
|
period_size = m_defPeriodSize;
|
|
dir = 1;
|
|
#if SND_LIB_MAJOR >= 1
|
|
err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &dir);
|
|
#else
|
|
err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, period_size, &dir);
|
|
#endif
|
|
if (err < 0) {
|
|
MSG("Setting period_size to %lu failed, but continuing: %s", period_size, snd_strerror(err));
|
|
}
|
|
|
|
periods = m_defPeriods;
|
|
dir = 1;
|
|
#if SND_LIB_MAJOR >= 1
|
|
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &periods, &dir);
|
|
#else
|
|
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, periods, &dir);
|
|
#endif
|
|
if (err < 0)
|
|
MSG("Unable to set number of periods to %i, but continuing: %s", periods, snd_strerror(err));
|
|
|
|
/* Install hw parameters. */
|
|
err = snd_pcm_hw_params(handle, hwparams);
|
|
if (err < 0) {
|
|
MSG("Unable to install hw params: %s", snd_strerror(err));
|
|
snd_pcm_hw_params_dump(hwparams, log);
|
|
stopAndExit();
|
|
}
|
|
|
|
/* Determine if device can pause. */
|
|
canPause = (1 == snd_pcm_hw_params_can_pause(hwparams));
|
|
|
|
/* Get final buffer size and calculate the chunk size we will pass to device. */
|
|
#if SND_LIB_MAJOR >= 1
|
|
snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
|
|
#else
|
|
buffer_size = snd_pcm_hw_params_get_buffer_size(hwparams);
|
|
#endif
|
|
chunk_size = periods * period_size;
|
|
|
|
if (0 == chunk_size) {
|
|
ERR("Invalid periods or period_size. Cannot continue.");
|
|
stopAndExit();
|
|
}
|
|
|
|
if (chunk_size == buffer_size)
|
|
MSG("WARNING: Shouldn't use chunk_size equal to buffer_size (%lu). Continuing anyway.", chunk_size);
|
|
|
|
DBG("Final buffer_size = %lu, chunk_size = %lu, periods = %i, period_size = %lu, canPause = %i",
|
|
buffer_size, chunk_size, periods, period_size, canPause);
|
|
|
|
if (m_debugLevel >= 2)
|
|
snd_pcm_dump(handle, log);
|
|
|
|
bits_per_sample = snd_pcm_format_physical_width(hwdata.format);
|
|
bits_per_frame = bits_per_sample * hwdata.channels;
|
|
chunk_bytes = chunk_size * bits_per_frame / 8;
|
|
audioBuffer.resize(chunk_bytes);
|
|
audiobuf = audioBuffer.data();
|
|
if (audiobuf == NULL) {
|
|
ERR("not enough memory");
|
|
stopAndExit();
|
|
}
|
|
}
|
|
|
|
#ifndef timersub
|
|
#define timersub(a, b, result) \
|
|
do { \
|
|
(result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
|
|
(result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
|
|
if ((result)->tv_usec < 0) { \
|
|
--(result)->tv_sec; \
|
|
(result)->tv_usec += 1000000; \
|
|
} \
|
|
} while (0)
|
|
#endif
|
|
|
|
/* I/O error handler */
|
|
void AlsaPlayer::xrun()
|
|
{
|
|
snd_pcm_status_t *status;
|
|
int res;
|
|
|
|
snd_pcm_status_alloca(&status);
|
|
if ((res = snd_pcm_status(handle, status))<0) {
|
|
ERR("status error: %s", snd_strerror(res));
|
|
stopAndExit();
|
|
}
|
|
if (SND_PCM_STATE_XRUN == snd_pcm_status_get_state(status)) {
|
|
struct timeval now, diff, tstamp;
|
|
gettimeofday(&now, 0);
|
|
snd_pcm_status_get_trigger_tstamp(status, &tstamp);
|
|
timersub(&now, &tstamp, &diff);
|
|
MSG("%s!!! (at least %.3f ms long)",
|
|
stream == SND_PCM_STREAM_PLAYBACK ? "underrun" : "overrun",
|
|
diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
|
|
if (m_debugLevel >= 2) {
|
|
DBG("Status:");
|
|
snd_pcm_status_dump(status, log);
|
|
}
|
|
if ((res = snd_pcm_prepare(handle))<0) {
|
|
ERR("xrun: prepare error: %s", snd_strerror(res));
|
|
stopAndExit();
|
|
}
|
|
return; /* ok, data should be accepted again */
|
|
} if (SND_PCM_STATE_DRAINING == snd_pcm_status_get_state(status)) {
|
|
if (m_debugLevel >= 2) {
|
|
DBG("Status(DRAINING):");
|
|
snd_pcm_status_dump(status, log);
|
|
}
|
|
if (stream == SND_PCM_STREAM_CAPTURE) {
|
|
MSG("capture stream format change? attempting recover...");
|
|
if ((res = snd_pcm_prepare(handle))<0) {
|
|
ERR("xrun(DRAINING): prepare error: %s", snd_strerror(res));
|
|
stopAndExit();
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
if (m_debugLevel >= 2) {
|
|
DBG("Status(R/W):");
|
|
snd_pcm_status_dump(status, log);
|
|
}
|
|
ERR("read/write error, state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status)));
|
|
stopAndExit();
|
|
}
|
|
|
|
/* I/O suspend handler */
|
|
void AlsaPlayer::suspend(void)
|
|
{
|
|
int res;
|
|
|
|
MSG("Suspended. Trying resume. ");
|
|
while ((res = snd_pcm_resume(handle)) == -EAGAIN)
|
|
sleep(1); /* wait until suspend flag is released */
|
|
if (res < 0) {
|
|
MSG("Failed. Restarting stream. ");
|
|
if ((res = snd_pcm_prepare(handle)) < 0) {
|
|
ERR("suspend: prepare error: %s", snd_strerror(res));
|
|
stopAndExit();
|
|
}
|
|
}
|
|
MSG("Suspend done.");
|
|
}
|
|
|
|
/* peak handler */
|
|
void AlsaPlayer::compute_max_peak(char *data, size_t count)
|
|
{
|
|
signed int val, max, max_peak = 0, perc;
|
|
size_t ocount = count;
|
|
|
|
switch (bits_per_sample) {
|
|
case 8: {
|
|
signed char *valp = (signed char *)data;
|
|
signed char mask = snd_pcm_format_silence(hwdata.format);
|
|
while (count-- > 0) {
|
|
val = *valp++ ^ mask;
|
|
val = abs(val);
|
|
if (max_peak < val)
|
|
max_peak = val;
|
|
}
|
|
break;
|
|
}
|
|
case 16: {
|
|
signed short *valp = (signed short *)data;
|
|
signed short mask = snd_pcm_format_silence_16(hwdata.format);
|
|
count /= 2;
|
|
while (count-- > 0) {
|
|
val = *valp++ ^ mask;
|
|
val = abs(val);
|
|
if (max_peak < val)
|
|
max_peak = val;
|
|
}
|
|
break;
|
|
}
|
|
case 32: {
|
|
signed int *valp = (signed int *)data;
|
|
signed int mask = snd_pcm_format_silence_32(hwdata.format);
|
|
count /= 4;
|
|
while (count-- > 0) {
|
|
val = *valp++ ^ mask;
|
|
val = abs(val);
|
|
if (max_peak < val)
|
|
max_peak = val;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
max = 1 << (bits_per_sample-1);
|
|
if (max <= 0)
|
|
max = 0x7fffffff;
|
|
DBG("Max peak (%li samples): %05i (0x%04x) ", (long)ocount, max_peak, max_peak);
|
|
if (bits_per_sample > 16)
|
|
perc = max_peak / (max / 100);
|
|
else
|
|
perc = max_peak * 100 / max;
|
|
for (val = 0; val < 20; val++)
|
|
if (val <= perc / 5)
|
|
kdDebug() << '#';
|
|
else
|
|
kdDebug() << ' ';
|
|
DBG(" %i%%", perc);
|
|
}
|
|
|
|
/*
|
|
* Write to the ALSA pcm.
|
|
*/
|
|
|
|
ssize_t AlsaPlayer::pcm_write(char *data, size_t count)
|
|
{
|
|
ssize_t r;
|
|
ssize_t result = 0;
|
|
|
|
if (sleep_min == 0 && count < chunk_size) {
|
|
DBG("calling snd_pcm_format_set_silence");
|
|
snd_pcm_format_set_silence(hwdata.format, data + count * bits_per_frame / 8, (chunk_size - count) * hwdata.channels);
|
|
count = chunk_size;
|
|
}
|
|
while (count > 0) {
|
|
DBG("calling writei_func, count = %i", count);
|
|
r = writei_func(handle, data, count);
|
|
DBG("writei_func returned %i", r);
|
|
if (-EAGAIN == r || (r >= 0 && (size_t)r < count)) {
|
|
DBG("r = %i calling snd_pcm_wait", r);
|
|
snd_pcm_wait(handle, 100);
|
|
} else if (-EPIPE == r) {
|
|
xrun();
|
|
} else if (-ESTRPIPE == r) {
|
|
suspend();
|
|
} else if (-EBUSY == r){
|
|
MSG("WARNING: sleeping while PCM BUSY");
|
|
usleep(1000);
|
|
continue;
|
|
} else if (r < 0) {
|
|
ERR("write error: %s", snd_strerror(r));
|
|
stopAndExit();
|
|
}
|
|
if (r > 0) {
|
|
if (m_debugLevel >= 1)
|
|
compute_max_peak(data, r * hwdata.channels);
|
|
result += r;
|
|
count -= r;
|
|
data += r * bits_per_frame / 8;
|
|
}
|
|
/* Report current state */
|
|
DBG("PCM state before polling: %s",
|
|
snd_pcm_state_name(snd_pcm_state(handle)));
|
|
|
|
int err = wait_for_poll(0);
|
|
if (err < 0) {
|
|
ERR("Wait for poll() failed");
|
|
return -1;
|
|
}
|
|
else if (err == 1){
|
|
MSG("Playback stopped");
|
|
/* Drop the playback on the sound device (probably
|
|
still in progress up till now) */
|
|
err = snd_pcm_drop(handle);
|
|
if (err < 0) {
|
|
ERR("snd_pcm_drop() failed: %s", snd_strerror(err));
|
|
return -1;
|
|
}
|
|
return -1;
|
|
}
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/*
|
|
* ok, let's play a .voc file
|
|
*/
|
|
|
|
ssize_t AlsaPlayer::voc_pcm_write(u_char *data, size_t count)
|
|
{
|
|
ssize_t result = count, r;
|
|
size_t size;
|
|
|
|
while (count > 0) {
|
|
size = count;
|
|
if (size > chunk_bytes - buffer_pos)
|
|
size = chunk_bytes - buffer_pos;
|
|
memcpy(audiobuf + buffer_pos, data, size);
|
|
data += size;
|
|
count -= size;
|
|
buffer_pos += size;
|
|
if ((size_t)buffer_pos == chunk_bytes) {
|
|
if ((size_t)(r = pcm_write(audiobuf, chunk_size)) != chunk_size)
|
|
return r;
|
|
buffer_pos = 0;
|
|
}
|
|
}
|
|
return result;
|
|
}
|
|
|
|
void AlsaPlayer::voc_write_silence(unsigned x)
|
|
{
|
|
unsigned l;
|
|
char *buf;
|
|
|
|
TQByteArray buffer(chunk_bytes);
|
|
// buf = (char *) malloc(chunk_bytes);
|
|
buf = buffer.data();
|
|
if (buf == NULL) {
|
|
ERR("can't allocate buffer for silence");
|
|
return; /* not fatal error */
|
|
}
|
|
snd_pcm_format_set_silence(hwdata.format, buf, chunk_size * hwdata.channels);
|
|
while (x > 0) {
|
|
l = x;
|
|
if (l > chunk_size)
|
|
l = chunk_size;
|
|
if (voc_pcm_write((u_char*)buf, l) != (ssize_t)l) {
|
|
ERR("write error");
|
|
stopAndExit();
|
|
}
|
|
x -= l;
|
|
}
|
|
// free(buf);
|
|
}
|
|
|
|
void AlsaPlayer::voc_pcm_flush(void)
|
|
{
|
|
if (buffer_pos > 0) {
|
|
size_t b;
|
|
if (sleep_min == 0) {
|
|
if (snd_pcm_format_set_silence(hwdata.format, audiobuf + buffer_pos, chunk_bytes - buffer_pos * 8 / bits_per_sample) < 0)
|
|
MSG("voc_pcm_flush - silence error");
|
|
b = chunk_size;
|
|
} else {
|
|
b = buffer_pos * 8 / bits_per_frame;
|
|
}
|
|
if (pcm_write(audiobuf, b) != (ssize_t)b)
|
|
ERR("voc_pcm_flush error");
|
|
}
|
|
snd_pcm_drain(handle);
|
|
}
|
|
|
|
void AlsaPlayer::voc_play(int fd, int ofs, const char* name)
|
|
{
|
|
int l;
|
|
VocBlockType *bp;
|
|
VocVoiceData *vd;
|
|
VocExtBlock *eb;
|
|
size_t nextblock, in_buffer;
|
|
u_char *data, *buf;
|
|
char was_extended = 0, output = 0;
|
|
u_short *sp, repeat = 0;
|
|
size_t silence;
|
|
off_t filepos = 0;
|
|
|
|
#define COUNT(x) nextblock -= x; in_buffer -= x; data += x
|
|
#define COUNT1(x) in_buffer -= x; data += x
|
|
|
|
TQByteArray buffer(64 * 1024);
|
|
// data = buf = (u_char *)malloc(64 * 1024);
|
|
data = buf = (u_char*)buffer.data();
|
|
buffer_pos = 0;
|
|
if (data == NULL) {
|
|
ERR("malloc error");
|
|
stopAndExit();
|
|
}
|
|
MSG("Playing Creative Labs Channel file '%s'...", name);
|
|
/* first we waste the rest of header, ugly but we don't need seek */
|
|
while (ofs > (ssize_t)chunk_bytes) {
|
|
if ((size_t)safe_read(fd, buf, chunk_bytes) != chunk_bytes) {
|
|
ERR("read error");
|
|
stopAndExit();
|
|
}
|
|
ofs -= chunk_bytes;
|
|
}
|
|
if (ofs) {
|
|
if (safe_read(fd, buf, ofs) != ofs) {
|
|
ERR("read error");
|
|
stopAndExit();
|
|
}
|
|
}
|
|
hwdata.format = DEFAULT_FORMAT;
|
|
hwdata.channels = 1;
|
|
hwdata.rate = DEFAULT_SPEED;
|
|
set_params();
|
|
|
|
in_buffer = nextblock = 0;
|
|
while (1) {
|
|
Fill_the_buffer: /* need this for repeat */
|
|
if (in_buffer < 32) {
|
|
/* move the rest of buffer to pos 0 and fill the buf up */
|
|
if (in_buffer)
|
|
memcpy(buf, data, in_buffer);
|
|
data = buf;
|
|
if ((l = safe_read(fd, buf + in_buffer, chunk_bytes - in_buffer)) > 0)
|
|
in_buffer += l;
|
|
else if (!in_buffer) {
|
|
/* the file is truncated, so simulate 'Terminator'
|
|
and reduce the datablock for safe landing */
|
|
nextblock = buf[0] = 0;
|
|
if (l == -1) {
|
|
// perror(name);
|
|
stopAndExit();
|
|
}
|
|
}
|
|
}
|
|
while (!nextblock) { /* this is a new block */
|
|
if (in_buffer < sizeof(VocBlockType))
|
|
goto __end;
|
|
bp = (VocBlockType *) data;
|
|
COUNT1(sizeof(VocBlockType));
|
|
nextblock = VOC_DATALEN(bp);
|
|
if (output)
|
|
MSG(" "); /* write /n after ASCII-out */
|
|
output = 0;
|
|
switch (bp->type) {
|
|
case 0:
|
|
#if 0
|
|
MSG("Terminator");
|
|
#endif
|
|
return; /* VOC-file stop */
|
|
case 1:
|
|
vd = (VocVoiceData *) data;
|
|
COUNT1(sizeof(VocVoiceData));
|
|
/* we need a SYNC, before we can set new SPEED, STEREO ... */
|
|
|
|
if (!was_extended) {
|
|
hwdata.rate = (int) (vd->tc);
|
|
hwdata.rate = 1000000 / (256 - hwdata.rate);
|
|
#if 0
|
|
MSG("Channel data %d Hz", dsp_speed);
|
|
#endif
|
|
if (vd->pack) { /* /dev/dsp can't it */
|
|
ERR("can't play packed .voc files");
|
|
return;
|
|
}
|
|
if (hwdata.channels == 2) /* if we are in Stereo-Mode, switch back */
|
|
hwdata.channels = 1;
|
|
} else { /* there was extended block */
|
|
hwdata.channels = 2;
|
|
was_extended = 0;
|
|
}
|
|
set_params();
|
|
break;
|
|
case 2: /* nothing to do, pure data */
|
|
#if 0
|
|
MSG("Channel continuation");
|
|
#endif
|
|
break;
|
|
case 3: /* a silence block, no data, only a count */
|
|
sp = (u_short *) data;
|
|
COUNT1(sizeof(u_short));
|
|
hwdata.rate = (int) (*data);
|
|
COUNT1(1);
|
|
hwdata.rate = 1000000 / (256 - hwdata.rate);
|
|
set_params();
|
|
silence = (((size_t) * sp) * 1000) / hwdata.rate;
|
|
#if 0
|
|
MSG("Silence for %d ms", (int) silence);
|
|
#endif
|
|
voc_write_silence(*sp);
|
|
break;
|
|
case 4: /* a marker for syncronisation, no effect */
|
|
sp = (u_short *) data;
|
|
COUNT1(sizeof(u_short));
|
|
#if 0
|
|
MSG("Marker %d", *sp);
|
|
#endif
|
|
break;
|
|
case 5: /* ASCII text, we copy to stderr */
|
|
output = 1;
|
|
#if 0
|
|
MSG("ASCII - text :");
|
|
#endif
|
|
break;
|
|
case 6: /* repeat marker, says repeatcount */
|
|
/* my specs don't say it: maybe this can be recursive, but
|
|
I don't think somebody use it */
|
|
repeat = *(u_short *) data;
|
|
COUNT1(sizeof(u_short));
|
|
#if 0
|
|
MSG("Repeat loop %d times", repeat);
|
|
#endif
|
|
if (filepos >= 0) { /* if < 0, one seek fails, why test another */
|
|
if ((filepos = lseek(fd, 0, 1)) < 0) {
|
|
ERR("can't play loops; %s isn't seekable", name);
|
|
repeat = 0;
|
|
} else {
|
|
filepos -= in_buffer; /* set filepos after repeat */
|
|
}
|
|
} else {
|
|
repeat = 0;
|
|
}
|
|
break;
|
|
case 7: /* ok, lets repeat that be rewinding tape */
|
|
if (repeat) {
|
|
if (repeat != 0xFFFF) {
|
|
#if 0
|
|
MSG("Repeat loop %d", repeat);
|
|
#endif
|
|
--repeat;
|
|
}
|
|
#if 0
|
|
else
|
|
MSG("Neverending loop");
|
|
#endif
|
|
lseek(fd, filepos, 0);
|
|
in_buffer = 0; /* clear the buffer */
|
|
goto Fill_the_buffer;
|
|
}
|
|
#if 0
|
|
else
|
|
MSG("End repeat loop");
|
|
#endif
|
|
break;
|
|
case 8: /* the extension to play Stereo, I have SB 1.0 :-( */
|
|
was_extended = 1;
|
|
eb = (VocExtBlock *) data;
|
|
COUNT1(sizeof(VocExtBlock));
|
|
hwdata.rate = (int) (eb->tc);
|
|
hwdata.rate = 256000000L / (65536 - hwdata.rate);
|
|
hwdata.channels = eb->mode == VOC_MODE_STEREO ? 2 : 1;
|
|
if (hwdata.channels == 2)
|
|
hwdata.rate = hwdata.rate >> 1;
|
|
if (eb->pack) { /* /dev/dsp can't it */
|
|
ERR("can't play packed .voc files");
|
|
return;
|
|
}
|
|
#if 0
|
|
MSG("Extended block %s %d Hz",
|
|
(eb->mode ? "Stereo" : "Mono"), dsp_speed);
|
|
#endif
|
|
break;
|
|
default:
|
|
ERR("unknown blocktype %d. terminate.", bp->type);
|
|
return;
|
|
} /* switch (bp->type) */
|
|
} /* while (! nextblock) */
|
|
/* put nextblock data bytes to dsp */
|
|
l = in_buffer;
|
|
if (nextblock < (size_t)l)
|
|
l = nextblock;
|
|
if (l) {
|
|
if (output) {
|
|
if (write(2, data, l) != l) { /* to stderr */
|
|
ERR("write error");
|
|
stopAndExit();
|
|
}
|
|
} else {
|
|
if (voc_pcm_write(data, l) != l) {
|
|
ERR("write error");
|
|
stopAndExit();
|
|
}
|
|
}
|
|
COUNT(l);
|
|
}
|
|
} /* while(1) */
|
|
__end:
|
|
voc_pcm_flush();
|
|
// free(buf);
|
|
}
|
|
/* that was a big one, perhaps somebody split it :-) */
|
|
|
|
/* setting the globals for playing raw data */
|
|
void AlsaPlayer::init_raw_data(void)
|
|
{
|
|
hwdata = rhwdata;
|
|
}
|
|
|
|
/* calculate the data count to read from/to dsp */
|
|
off_t AlsaPlayer::calc_count(void)
|
|
{
|
|
off_t count;
|
|
|
|
if (timelimit == 0) {
|
|
count = pbrec_count;
|
|
} else {
|
|
count = snd_pcm_format_size(hwdata.format, hwdata.rate * hwdata.channels);
|
|
count *= (off_t)timelimit;
|
|
}
|
|
return count < pbrec_count ? count : pbrec_count;
|
|
}
|
|
|
|
void AlsaPlayer::header(int /*rtype*/, const char* /*name*/)
|
|
{
|
|
// fprintf(stderr, "%s %s '%s' : ",
|
|
// (stream == SND_PCM_STREAM_PLAYBACK) ? "Playing" : "Recording",
|
|
// fmt_rec_table[rtype].what,
|
|
// name);
|
|
TQString channels;
|
|
if (hwdata.channels == 1)
|
|
channels = "Mono";
|
|
else if (hwdata.channels == 2)
|
|
channels = "Stereo";
|
|
else
|
|
channels = TQString("Channels %1").arg(hwdata.channels);
|
|
DBG("Format: %s, Rate %d Hz, %s",
|
|
snd_pcm_format_description(hwdata.format),
|
|
hwdata.rate,
|
|
channels.ascii());
|
|
}
|
|
|
|
/* playing raw data */
|
|
|
|
void AlsaPlayer::playback_go(int fd, size_t loaded, off_t count, int rtype, const char *name)
|
|
{
|
|
int l, r;
|
|
off_t written = 0;
|
|
off_t c;
|
|
|
|
if (m_debugLevel >= 1) header(rtype, name);
|
|
set_params();
|
|
|
|
while (loaded > chunk_bytes && written < count) {
|
|
if (pcm_write(audiobuf + written, chunk_size) <= 0)
|
|
return;
|
|
written += chunk_bytes;
|
|
loaded -= chunk_bytes;
|
|
}
|
|
if (written > 0 && loaded > 0)
|
|
memmove(audiobuf, audiobuf + written, loaded);
|
|
|
|
l = loaded;
|
|
while (written < count) {
|
|
do {
|
|
c = count - written;
|
|
if (c > chunk_bytes)
|
|
c = chunk_bytes;
|
|
c -= l;
|
|
|
|
if (c == 0)
|
|
break;
|
|
r = safe_read(fd, audiobuf + l, c);
|
|
if (r < 0) {
|
|
// perror(name);
|
|
stopAndExit();
|
|
}
|
|
fdcount += r;
|
|
if (r == 0)
|
|
break;
|
|
l += r;
|
|
} while (sleep_min == 0 && (size_t)l < chunk_bytes);
|
|
l = l * 8 / bits_per_frame;
|
|
DBG("calling pcm_write with %i frames.", l);
|
|
r = pcm_write(audiobuf, l);
|
|
DBG("pcm_write returned r = %i", r);
|
|
if (r < 0) return;
|
|
if (r != l)
|
|
break;
|
|
r = r * bits_per_frame / 8;
|
|
written += r;
|
|
l = 0;
|
|
}
|
|
|
|
DBG("Draining...");
|
|
|
|
/* We want the next "device ready" notification only when the buffer is completely empty. */
|
|
/* Do this by setting the avail_min to the buffer size. */
|
|
int err;
|
|
DBG("Getting swparams");
|
|
snd_pcm_sw_params_t *swparams;
|
|
snd_pcm_sw_params_alloca(&swparams);
|
|
err = snd_pcm_sw_params_current(handle, swparams);
|
|
if (err < 0) {
|
|
ERR("Unable to get current swparams: %s", snd_strerror(err));
|
|
return;
|
|
}
|
|
DBG("Setting avail min to %lu", buffer_size);
|
|
err = snd_pcm_sw_params_set_avail_min(handle, swparams, buffer_size);
|
|
if (err < 0) {
|
|
ERR("Unable to set avail min for playback: %s", snd_strerror(err));
|
|
return;
|
|
}
|
|
/* write the parameters to the playback device */
|
|
DBG("Writing swparams");
|
|
err = snd_pcm_sw_params(handle, swparams);
|
|
if (err < 0) {
|
|
ERR("Unable to set sw params for playback: %s", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
DBG("Waiting for poll");
|
|
err = wait_for_poll(1);
|
|
if (err < 0) {
|
|
ERR("Wait for poll() failed");
|
|
return;
|
|
} else if (err == 1){
|
|
MSG("Playback stopped while draining");
|
|
|
|
/* Drop the playback on the sound device (probably
|
|
still in progress up till now) */
|
|
err = snd_pcm_drop(handle);
|
|
if (err < 0) {
|
|
ERR("snd_pcm_drop() failed: %s", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
DBG("Draining completed");
|
|
}
|
|
|
|
/*
|
|
* let's play or capture it (capture_type says VOC/WAVE/raw)
|
|
*/
|
|
|
|
void AlsaPlayer::playback(int fd)
|
|
{
|
|
int ofs;
|
|
size_t dta;
|
|
ssize_t dtawave;
|
|
|
|
pbrec_count = LLONG_MAX;
|
|
fdcount = 0;
|
|
|
|
/* read the file header */
|
|
dta = sizeof(AuHeader);
|
|
if ((size_t)safe_read(fd, audiobuf, dta) != dta) {
|
|
ERR("read error");
|
|
stopAndExit();
|
|
}
|
|
if (test_au(fd, audiobuf) >= 0) {
|
|
rhwdata.format = hwdata.format;
|
|
pbrec_count = calc_count();
|
|
playback_go(fd, 0, pbrec_count, FORMAT_AU, name.ascii());
|
|
goto __end;
|
|
}
|
|
dta = sizeof(VocHeader);
|
|
if ((size_t)safe_read(fd, audiobuf + sizeof(AuHeader),
|
|
dta - sizeof(AuHeader)) != dta - sizeof(AuHeader)) {
|
|
ERR("read error");
|
|
stopAndExit();
|
|
}
|
|
if ((ofs = test_vocfile(audiobuf)) >= 0) {
|
|
pbrec_count = calc_count();
|
|
voc_play(fd, ofs, name.ascii());
|
|
goto __end;
|
|
}
|
|
/* read bytes for WAVE-header */
|
|
if ((dtawave = test_wavefile(fd, audiobuf, dta)) >= 0) {
|
|
pbrec_count = calc_count();
|
|
playback_go(fd, dtawave, pbrec_count, FORMAT_WAVE, name.ascii());
|
|
} else {
|
|
/* should be raw data */
|
|
init_raw_data();
|
|
pbrec_count = calc_count();
|
|
playback_go(fd, dta, pbrec_count, FORMAT_RAW, name.ascii());
|
|
}
|
|
__end:
|
|
return;
|
|
}
|
|
|
|
/* Wait until ALSA is ready for more samples or stop() was called.
|
|
@return 0 if ALSA is ready for more input, +1 if a request to stop
|
|
the sound output was received and a negative value on error. */
|
|
int AlsaPlayer::wait_for_poll(int draining)
|
|
{
|
|
unsigned short revents;
|
|
snd_pcm_state_t state;
|
|
int ret;
|
|
|
|
DBG("Waiting for poll");
|
|
|
|
/* Wait for certain events */
|
|
while (1) {
|
|
/* Simulated pause by not writing to alsa device, which will lead to an XRUN
|
|
when resumed. */
|
|
if (m_simulatedPause)
|
|
msleep(500);
|
|
else {
|
|
|
|
ret = poll(alsa_poll_fds, alsa_fd_count, -1);
|
|
DBG("activity on %d descriptors", ret);
|
|
|
|
/* Check for stop request from alsa_stop on the last file descriptors. */
|
|
if ((revents = alsa_poll_fds[alsa_fd_count-1].revents)) {
|
|
if (revents & POLLIN){
|
|
DBG("stop requested");
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
/* Check the first count-1 descriptors for ALSA events */
|
|
snd_pcm_poll_descriptors_revents(handle, alsa_poll_fds, alsa_fd_count-1, &revents);
|
|
|
|
/* Ensure we are in the right state */
|
|
state = snd_pcm_state(handle);
|
|
DBG("State after poll returned is %s", snd_pcm_state_name(state));
|
|
|
|
if (SND_PCM_STATE_XRUN == state){
|
|
if (!draining){
|
|
MSG("WARNING: Buffer underrun detected!");
|
|
xrun();
|
|
return 0;
|
|
}else{
|
|
DBG("Playback terminated");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (SND_PCM_STATE_SUSPENDED == state){
|
|
DBG("WARNING: Suspend detected!");
|
|
suspend();
|
|
return 0;
|
|
}
|
|
|
|
/* Check for errors */
|
|
if (revents & POLLERR) {
|
|
DBG("poll revents says POLLERR");
|
|
return -EIO;
|
|
}
|
|
|
|
/* Is ALSA ready for more input? */
|
|
if ((revents & POLLOUT)){
|
|
DBG("Ready for more input");
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#include "alsaplayer.moc"
|
|
|
|
#undef DBG
|
|
#undef MSG
|
|
#undef ERR
|