diff options
Diffstat (limited to 'kttsd/players/alsaplayer/alsaplayer.cpp')
| -rw-r--r-- | kttsd/players/alsaplayer/alsaplayer.cpp | 1729 |
1 files changed, 1729 insertions, 0 deletions
diff --git a/kttsd/players/alsaplayer/alsaplayer.cpp b/kttsd/players/alsaplayer/alsaplayer.cpp new file mode 100644 index 0000000..d3eabe8 --- /dev/null +++ b/kttsd/players/alsaplayer/alsaplayer.cpp @@ -0,0 +1,1729 @@ +/***************************************************** vim:set ts=4 sw=4 sts=4: + ALSA player. + ------------------- + Copyright: + (C) 2005 by Gary Cramblitt <garycramblitt@comcast.net> + Portions based on aplay.c in alsa-utils + Copyright (c) by Jaroslav Kysela <perex@suse.cz> + Based on vplay program by Michael Beck + ------------------- + Original author: Gary Cramblitt <garycramblitt@comcast.net> + + 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 <sys/wait.h> +// System includes. +#include <config.h> +#if TIME_WITH_SYS_TIME +# include <sys/time.h> +# include <time.h> +#else +# if HAVE_SYS_TIME_H +# include <sys/time.h> +# else +# include <time.h> +# endif +#endif + +// Qt includes. +#include <qdir.h> +#include <qapplication.h> +#include <qcstring.h> + +// KDE includes. +#include <kdebug.h> +#include <kconfig.h> +#include <kstandarddirs.h> +#include <kmessagebox.h> +#include <klocale.h> + +// AlsaPlayer includes. +#include "alsaplayer.h" + +#if !defined(__GNUC__) || __GNUC__ >= 3 +#define ERR(...) do {\ + QString dbgStr;\ + QString 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 {\ + QString dbgStr;\ + QString 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) {\ + QString dbgStr; \ + QString 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) {\ + QString dbgStr; \ + QString 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) {\ + QString dbgStr; \ + QString 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) {\ + QString dbgStr; \ + QString s = dbgStr.sprintf( "%s:%d: ", __FUNCTION__, __LINE__); \ + s += dbgStr.sprintf( ##args ); \ + kdDebug() << timestamp() << "AlsaPlayer::" << s << endl; \ + }; \ +} while (0) +#endif + +QString 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); + QString ts; + ts.sprintf(" %s [%d] ",tstr, (int) tv.tv_usec); + free(tstr); + return ts; +} + +//////////////////////////////////////////////////////////////////////////////// +// public methods +//////////////////////////////////////////////////////////////////////////////// + +AlsaPlayer::AlsaPlayer(QObject* parent, const char* name, const QStringList& 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 QString &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() +{ + QString pName = m_pcmName.section(" ", 0, 0); + DBG("pName = %s", pName.ascii()); + pcm_name = qstrdup(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). + */ +QStringList AlsaPlayer::getPluginList( const QCString& /*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); + QStringList 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; + QString infoName = " "; + infoName += snd_ctl_card_info_get_name(info); + infoName += " ("; + infoName += snd_pcm_info_get_name(pcminfo); + infoName += ")"; + if (0 == devCnt) { + QString pcmName = QString("default:%1").arg(card); + result.append(pcmName + infoName); + } + QString pcmName = QString("plughw:%1,%2").arg(card).arg(device); + result.append(pcmName + infoName); + } + snd_ctl_close(handle); + } + } + return result; +} + +// QStringList AlsaPlayer::getPluginList( const QCString& /*classname*/ ) +// { +// QStringList 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 QStringList(); +// 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; +// QStringList 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 (QString(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) { +// QString pluginName = name; +// pluginName += ","; +// pluginName += QString::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 QString& 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; + + QByteArray 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; + off64_t filepos = 0; + +#define COUNT(x) nextblock -= x; in_buffer -= x; data += x +#define COUNT1(x) in_buffer -= x; data += x + + QByteArray 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 = lseek64(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 + lseek64(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 */ +off64_t AlsaPlayer::calc_count(void) +{ + off64_t count; + + if (timelimit == 0) { + count = pbrec_count; + } else { + count = snd_pcm_format_size(hwdata.format, hwdata.rate * hwdata.channels); + count *= (off64_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); + QString channels; + if (hwdata.channels == 1) + channels = "Mono"; + else if (hwdata.channels == 2) + channels = "Stereo"; + else + channels = QString("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, off64_t count, int rtype, const char *name) +{ + int l, r; + off64_t written = 0; + off64_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 |