Aegisub/aegisub/src/audio_player_oss.cpp
Thomas Goyne 7218c04d52 Kill AudioPlayer::displayTimer
The rewritten audio display does not use displayTimer and that
functionality shouldn't have been in the players in the first place.

Originally committed to SVN as r6605.
2012-03-25 04:05:44 +00:00

252 lines
6.8 KiB
C++

// Copyright (c) 2009, Grigori Goronzy
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// Aegisub Project http://www.aegisub.org/
//
// $Id$
/// @file audio_player_oss.cpp
/// @brief Open Sound System audio output
/// @ingroup audio_output
///
#include "config.h"
#ifdef WITH_OSS
#ifndef AGI_PRE
#include <sys/param.h>
#endif
#include <libaegisub/log.h>
#include "audio_player_oss.h"
#include "audio_controller.h"
#include "compat.h"
#include "include/aegisub/audio_provider.h"
#include "main.h"
#include "utils.h"
DEFINE_SIMPLE_EXCEPTION(OSSError, agi::AudioPlayerOpenError, "audio/player/open/oss")
OSSPlayer::OSSPlayer()
{
volume = 1.0f;
open = false;
playing = false;
start_frame = cur_frame = end_frame = bpf = 0;
provider = 0;
thread = 0;
}
OSSPlayer::~OSSPlayer()
{
CloseStream();
}
void OSSPlayer::OpenStream()
{
CloseStream();
bpf = provider->GetChannels() * provider->GetBytesPerSample();
// Open device
wxString device = lagi_wxString(OPT_GET("Player/Audio/OSS/Device")->GetString());
dspdev = ::open(device.mb_str(wxConvUTF8), O_WRONLY, 0);
if (dspdev < 0) {
throw OSSError("OSS player: opening device failed", 0);
}
// Use a reasonable buffer policy for low latency (OSS4)
#ifdef SNDCTL_DSP_POLICY
int policy = 3;
ioctl(dspdev, SNDCTL_DSP_POLICY, &policy);
#endif
// Set number of channels
int channels = provider->GetChannels();
if (ioctl(dspdev, SNDCTL_DSP_CHANNELS, &channels) < 0) {
throw OSSError("OSS player: setting channels failed", 0);
}
// Set sample format
int sample_format;
switch (provider->GetBytesPerSample()) {
case 1:
sample_format = AFMT_S8;
break;
case 2:
sample_format = AFMT_S16_LE;
break;
default:
throw OSSError("OSS player: can only handle 8 and 16 bit sound", 0);
}
if (ioctl(dspdev, SNDCTL_DSP_SETFMT, &sample_format) < 0) {
throw OSSError("OSS player: setting sample format failed", 0);
}
// Set sample rate
rate = provider->GetSampleRate();
if (ioctl(dspdev, SNDCTL_DSP_SPEED, &rate) < 0) {
throw OSSError("OSS player: setting samplerate failed", 0);
}
// Now ready
open = true;
}
void OSSPlayer::CloseStream()
{
if (!open) return;
Stop();
::close(dspdev);
// No longer working
open = false;
}
void OSSPlayer::Play(int64_t start, int64_t count)
{
Stop();
start_frame = cur_frame = start;
end_frame = start + count;
thread = new OSSPlayerThread(this);
thread->Create();
thread->Run();
playing = true;
}
void OSSPlayer::Stop()
{
if (!open) return;
if (!playing) return;
// Stop the thread
if (thread) {
if (thread->IsAlive()) {
thread->Delete();
}
thread->Wait();
delete thread;
}
// errors can be ignored here
ioctl(dspdev, SNDCTL_DSP_RESET, NULL);
// Reset data
playing = false;
start_frame = 0;
cur_frame = 0;
end_frame = 0;
}
void OSSPlayer::SetEndPosition(int64_t pos)
{
end_frame = pos;
if (pos <= GetCurrentPosition()) {
ioctl(dspdev, SNDCTL_DSP_RESET, NULL);
if (thread && thread->IsAlive())
thread->Delete();
}
}
void OSSPlayer::SetCurrentPosition(int64_t pos)
{
cur_frame = start_frame = pos;
}
int64_t OSSPlayer::GetCurrentPosition()
{
if (!playing)
return 0;
#ifdef SNDCTL_DSP_CURRENT_OPTR
// OSS4
long played_frames = 0;
oss_count_t pos;
if (ioctl(dspdev, SNDCTL_DSP_CURRENT_OPTR, &pos) >= 0) {
// XXX: Apparently the semantics are different on FreeBSD...
#ifdef __FREEBSD__
played_frames = MAX(0, pos.samples - pos.fifo_samples);
#else
played_frames = pos.samples + pos.fifo_samples;
#endif
LOG_D("player/audio/oss") << "played_frames: " << played_frames << " fifo " << pos.fifo_samples;
return start_frame + played_frames;
}
#endif
// Fallback for old OSS versions
int delay = 0;
if (ioctl(dspdev, SNDCTL_DSP_GETODELAY, &delay) >= 0) {
delay /= bpf;
LOG_D("player/audio/oss") << "cur_frame: " << cur_frame << " delay " << delay;
// delay can jitter a bit at the end, detect that
if (cur_frame == end_frame && delay < rate / 20) {
return cur_frame;
}
return MAX(0, (long) cur_frame - delay);
}
// Maybe this still didn't work...
// Return the last written frame, timing will suffer
return cur_frame;
}
OSSPlayerThread::OSSPlayerThread(OSSPlayer *par)
: wxThread(wxTHREAD_JOINABLE)
, parent(par)
{
}
wxThread::ExitCode OSSPlayerThread::Entry() {
// Use small enough writes for good timing accuracy with all
// timing methods.
const unsigned long wsize = parent->rate / 25;
void *buf = malloc(wsize * parent->bpf);
while (!TestDestroy() && parent->cur_frame < parent->end_frame) {
int rsize = std::min(wsize, parent->end_frame - parent->cur_frame);
parent->provider->GetAudioWithVolume(buf, parent->cur_frame,
rsize, parent->volume);
int written = ::write(parent->dspdev, buf, rsize * parent->bpf);
parent->cur_frame += written / parent->bpf;
}
free(buf);
parent->cur_frame = parent->end_frame;
LOG_D("player/audio/oss") << "Thread dead";
return 0;
}
#endif // WITH_OSS