/***************************************************** vim:set ts=4 sw=4 sts=4: ALSA player. ------------------- Copyright: (C) 2005 by Gary Cramblitt Portions based on aplay.c in alsa-utils Copyright (c) by Jaroslav Kysela Based on vplay program by Michael Beck ------------------- Original author: Gary Cramblitt 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. ******************************************************************************/ // #include // System includes. #include #if TIME_WITH_SYS_TIME # include # include #else # if HAVE_SYS_TIME_H # include # else # include # endif #endif // TQt includes. #include #include #include // KDE includes. #include #include #include #include #include // AlsaPlayer includes. #include "alsaplayer.h" #if !defined(__GNUC__) || __GNUC__ >= 3 #define ERR(...) do {\ TQString dbgStr;\ TQString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( __VA_ARGS__); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ } while (0) #else #define ERR(args...) do {\ TQString dbgStr;\ TQString s = dbgStr.sprintf( "%s:%d: ERROR ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( ##args ); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ } while (0) #endif #if !defined(__GNUC__) || __GNUC__ >= 3 #define MSG(...) do {\ if (m_debugLevel >= 1) {\ TQString dbgStr; \ TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( __VA_ARGS__); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ }; \ } while (0) #else #define MSG(args...) do {\ if (m_debugLevel >= 1) {\ TQString dbgStr; \ TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( ##args ); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ }; \ } while (0) #endif #if !defined(__GNUC__) || __GNUC__ >= 3 #define DBG(...) do {\ if (m_debugLevel >= 2) {\ TQString dbgStr; \ TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( __VA_ARGS__); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ }; \ } while (0) #else #define DBG(args...) do {\ if (m_debugLevel >= 2) {\ TQString dbgStr; \ TQString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \ s += dbgStr.sprintf( ##args ); \ kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ }; \ } while (0) #endif TQString AlsaPlayer::timestamp() const { time_t t; struct timeval tv; char *tstr; t = time(NULL); tstr = strdup(ctime(&t)); tstr[strlen(tstr)-1] = 0; gettimeofday(&tv,NULL); TQString ts; ts.sprintf(" %s [%d] ",tstr, (int) tv.tv_usec); free(tstr); return ts; } //////////////////////////////////////////////////////////////////////////////// // public methods //////////////////////////////////////////////////////////////////////////////// AlsaPlayer::AlsaPlayer(TQObject* parent, const char* name, const TQStringList& args) : Player(parent, name, args), m_currentVolume(1.0), m_pcmName("default"), m_defPeriodSize(128), m_defPeriods(8), m_debugLevel(1), m_simulatedPause(false) { init(); } AlsaPlayer::~AlsaPlayer() { if (running()) { stop(); wait(); } } //void AlsaPlayer::play(const FileHandle &file) void AlsaPlayer::startPlay(const TQString &file) { if (running()) { if (paused()) { if (canPause) snd_pcm_pause(handle, false); else m_simulatedPause = false; } return; } audiofile.setName(file); audiofile.open(IO_ReadOnly); fd = audiofile.handle(); // Start thread running. start(); } /*virtual*/ void AlsaPlayer::run() { TQString pName = m_pcmName.section(" ", 0, 0); DBG("pName = %s", pName.ascii()); pcm_name = tqstrdup(pName.ascii()); int err; snd_pcm_info_t *info; m_simulatedPause = false; snd_pcm_info_alloca(&info); err = snd_output_stdio_attach(&log, stderr, 0); assert(err >= 0); rhwdata.format = DEFAULT_FORMAT; rhwdata.rate = DEFAULT_SPEED; rhwdata.channels = 1; err = snd_pcm_open(&handle, pcm_name, stream, open_mode); if (err < 0) { ERR("audio open error on pcm device %s: %s", pcm_name, snd_strerror(err)); return; } if ((err = snd_pcm_info(handle, info)) < 0) { ERR("info error: %s", snd_strerror(err)); return; } chunk_size = 1024; hwdata = rhwdata; audioBuffer.resize(1024); // audiobuf = (char *)malloc(1024); audiobuf = audioBuffer.data(); if (audiobuf == NULL) { ERR("not enough memory"); return; } if (mmap_flag) { writei_func = snd_pcm_mmap_writei; readi_func = snd_pcm_mmap_readi; writen_func = snd_pcm_mmap_writen; readn_func = snd_pcm_mmap_readn; } else { writei_func = snd_pcm_writei; readi_func = snd_pcm_readi; writen_func = snd_pcm_writen; readn_func = snd_pcm_readn; } playback(fd); cleanup(); return; } void AlsaPlayer::pause() { if (running()) { DBG("Pause requested"); m_mutex.lock(); if (handle) { // Some hardware can pause; some can't. canPause is set in set_params. if (canPause) { m_simulatedPause = false; snd_pcm_pause(handle, true); m_mutex.unlock(); } else { // Set a flag and cause wait_for_poll to sleep. When resumed, will get // an underrun. m_simulatedPause = true; m_mutex.unlock(); } } } } void AlsaPlayer::stop() { if (running()) { DBG("STOP! Locking mutex"); m_mutex.lock(); m_simulatedPause = false; if (handle) { /* This constant is arbitrary */ char buf = 42; DBG("Request for stop, device state is %s", snd_pcm_state_name(snd_pcm_state(handle))); write(alsa_stop_pipe[1], &buf, 1); } DBG("unlocking mutex"); m_mutex.unlock(); /* Wait for thread to exit */ DBG("waiting for thread to exit"); wait(); DBG("cleaning up"); } cleanup(); } /* * Stop playback, cleanup and exit thread. */ void AlsaPlayer::stopAndExit() { // if (handle) snd_pcm_drop(handle); cleanup(); exit(); } void AlsaPlayer::setVolume(float volume) { m_currentVolume = volume; } float AlsaPlayer::volume() const { return m_currentVolume; } ///////////////////////////////////////////////////////////////////////////////// // player status functions ///////////////////////////////////////////////////////////////////////////////// bool AlsaPlayer::playing() const { bool result = false; if (running()) { m_mutex.lock(); if (handle) { if (canPause) { snd_pcm_status_t *status; snd_pcm_status_alloca(&status); int res; if ((res = snd_pcm_status(handle, status)) < 0) ERR("status error: %s", snd_strerror(res)); else { result = (SND_PCM_STATE_RUNNING == snd_pcm_status_get_state(status)) || (SND_PCM_STATE_DRAINING == snd_pcm_status_get_state(status)); DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status))); } } else result = !m_simulatedPause; } m_mutex.unlock(); } return result; } bool AlsaPlayer::paused() const { bool result = false; if (running()) { m_mutex.lock(); if (handle) { if (canPause) { snd_pcm_status_t *status; snd_pcm_status_alloca(&status); int res; if ((res = snd_pcm_status(handle, status)) < 0) ERR("status error: %s", snd_strerror(res)); else { result = (SND_PCM_STATE_PAUSED == snd_pcm_status_get_state(status)); DBG("state = %s", snd_pcm_state_name(snd_pcm_status_get_state(status))); } } else result = m_simulatedPause; } m_mutex.unlock(); } return result; } int AlsaPlayer::totalTime() const { int total = 0; int rate = hwdata.rate; int channels = hwdata.channels; if (rate > 0 && channels > 0) { total = int((double(pbrec_count) / rate) / channels); // DBG("pbrec_count = %i rate =%i channels = %i", pbrec_count, rate, channels); // DBG("totalTime = %i", total); } return total; } int AlsaPlayer::currentTime() const { int current = 0; int rate = hwdata.rate; int channels = hwdata.channels; if (rate > 0 && channels > 0) { current = int((double(fdcount) / rate) / channels); // DBG("fdcount = %i rate = %i channels = %i", fdcount, rate, channels); // DBG("currentTime = %i", current); } return current; } int AlsaPlayer::position() const { // TODO: Make this more accurate by adding frames that have been so-far // played within the Alsa ring buffer. return pbrec_count > 0 ? int(double(fdcount) * 1000 / pbrec_count + .5) : 0; } ///////////////////////////////////////////////////////////////////////////////// // player seek functions ///////////////////////////////////////////////////////////////////////////////// void AlsaPlayer::seek(int /*seekTime*/) { // TODO: } void AlsaPlayer::seekPosition(int /*position*/) { // TODO: } /* * Returns a list of PCM devices. * This function fills the specified list with ALSA hardware soundcards found on the system. * It uses plughw:xx instead of hw:xx for specifiers, because hw:xx are not practical to * use (e.g. they require a resampler/channel mixer in the application). */ TQStringList AlsaPlayer::getPluginList( const TQCString& /*classname*/ ) { int err = 0; int card = -1, device = -1; snd_ctl_t *handle; snd_ctl_card_info_t *info; snd_pcm_info_t *pcminfo; snd_ctl_card_info_alloca(&info); snd_pcm_info_alloca(&pcminfo); TQStringList result; result.append("default"); for (;;) { err = snd_card_next(&card); if (err < 0 || card < 0) break; if (card >= 0) { char name[32]; sprintf(name, "hw:%i", card); if ((err = snd_ctl_open(&handle, name, 0)) < 0) continue; if ((err = snd_ctl_card_info(handle, info)) < 0) { snd_ctl_close(handle); continue; } for (int devCnt=0;;++devCnt) { err = snd_ctl_pcm_next_device(handle, &device); if (err < 0 || device < 0) break; snd_pcm_info_set_device(pcminfo, device); snd_pcm_info_set_subdevice(pcminfo, 0); snd_pcm_info_set_stream(pcminfo, SND_PCM_STREAM_PLAYBACK); if ((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) continue; TQString infoName = " "; infoName += snd_ctl_card_info_get_name(info); infoName += " ("; infoName += snd_pcm_info_get_name(pcminfo); infoName += ")"; if (0 == devCnt) { TQString pcmName = TQString("default:%1").arg(card); result.append(pcmName + infoName); } TQString pcmName = TQString("plughw:%1,%2").arg(card).arg(device); result.append(pcmName + infoName); } snd_ctl_close(handle); } } return result; } // TQStringList AlsaPlayer::getPluginList( const TQCString& /*classname*/ ) // { // TQStringList assumed("default"); // snd_config_t *conf; // int err = snd_config_update(); // if (err < 0) { // 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 // vim: sw=4 ts=8 et