4a8abae3a3
Originally committed to SVN as r4676.
347 lines
7.8 KiB
C++
347 lines
7.8 KiB
C++
// Copyright (c) 2009, Grigori Goronzy
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// Aegisub Project http://www.aegisub.org/
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//
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// $Id$
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/// @file audio_player_oss.cpp
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/// @brief Open Sound System audio output
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/// @ingroup audio_output
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///
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#include "config.h"
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#ifdef WITH_OSS
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///////////
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// Headers
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#include <libaegisub/log.h>
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#include "audio_player_manager.h"
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#include "audio_player_oss.h"
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#include "audio_provider_manager.h"
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#include "frame_main.h"
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#include "compat.h"
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#include "main.h"
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#include "utils.h"
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/// @brief Constructor
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///
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OSSPlayer::OSSPlayer()
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{
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volume = 1.0f;
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open = false;
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playing = false;
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start_frame = cur_frame = end_frame = bpf = 0;
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provider = 0;
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thread = 0;
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}
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/// @brief Destructor
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///
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OSSPlayer::~OSSPlayer()
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{
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CloseStream();
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}
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/// @brief Open stream
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///
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void OSSPlayer::OpenStream()
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{
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CloseStream();
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// Get provider
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provider = GetProvider();
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bpf = provider->GetChannels() * provider->GetBytesPerSample();
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// Open device
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wxString device = lagi_wxString(OPT_GET("Audio/OSS/Device")->GetString());
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dspdev = ::open(device.mb_str(wxConvUTF8), O_WRONLY, 0);
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if (dspdev < 0) {
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throw _T("OSS player: opening device failed");
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}
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// Use a reasonable buffer policy for low latency (OSS4)
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#ifdef SNDCTL_DSP_POLICY
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int policy = 3;
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ioctl(dspdev, SNDCTL_DSP_POLICY, &policy);
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#endif
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// Set number of channels
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int channels = provider->GetChannels();
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if (ioctl(dspdev, SNDCTL_DSP_CHANNELS, &channels) < 0) {
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throw _T("OSS player: setting channels failed");
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}
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// Set sample format
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int sample_format;
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switch (provider->GetBytesPerSample()) {
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case 1:
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sample_format = AFMT_S8;
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break;
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case 2:
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sample_format = AFMT_S16_LE;
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break;
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default:
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throw _T("OSS player: can only handle 8 and 16 bit sound");
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}
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if (ioctl(dspdev, SNDCTL_DSP_SETFMT, &sample_format) < 0) {
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throw _T("OSS player: setting sample format failed");
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}
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// Set sample rate
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rate = provider->GetSampleRate();
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if (ioctl(dspdev, SNDCTL_DSP_SPEED, &rate) < 0) {
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throw("OSS player: setting samplerate failed");
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}
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// Now ready
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open = true;
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}
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/// @brief Close stream
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/// @return
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///
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void OSSPlayer::CloseStream()
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{
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if (!open) return;
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Stop();
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::close(dspdev);
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// No longer working
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open = false;
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}
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/// @brief Play
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/// @param start
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/// @param count
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///
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void OSSPlayer::Play(int64_t start, int64_t count)
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{
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Stop();
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start_frame = cur_frame = start;
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end_frame = start + count;
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thread = new OSSPlayerThread(this);
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thread->Create();
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thread->Run();
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// Update timer
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if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15);
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playing = true;
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}
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/// @brief Stop
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/// @param timerToo
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/// @return
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///
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void OSSPlayer::Stop(bool timerToo)
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{
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if (!open) return;
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if (!playing) return;
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// Stop the thread
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if (thread) {
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if (thread->IsAlive()) {
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thread->Delete();
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}
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thread->Wait();
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delete thread;
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}
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// errors can be ignored here
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ioctl(dspdev, SNDCTL_DSP_RESET, NULL);
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// Reset data
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playing = false;
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start_frame = 0;
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cur_frame = 0;
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end_frame = 0;
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// Stop timer
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if (timerToo && displayTimer) {
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displayTimer->Stop();
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}
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}
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/// @brief DOCME
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/// @return
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///
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bool OSSPlayer::IsPlaying()
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{
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return playing;
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}
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/// @brief Set end
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/// @param pos
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///
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void OSSPlayer::SetEndPosition(int64_t pos)
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{
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end_frame = pos;
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if (end_frame <= GetCurrentPosition()) {
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ioctl(dspdev, SNDCTL_DSP_RESET, NULL);
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if (thread && thread->IsAlive())
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thread->Delete();
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}
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}
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/// @brief Set current position
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/// @param pos
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///
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void OSSPlayer::SetCurrentPosition(int64_t pos)
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{
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cur_frame = start_frame = pos;
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}
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/// @brief DOCME
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/// @return
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///
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int64_t OSSPlayer::GetStartPosition()
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{
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return start_frame;
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}
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/// @brief DOCME
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/// @return
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///
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int64_t OSSPlayer::GetEndPosition()
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{
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return end_frame;
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}
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/// @brief Get current position
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/// @return
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///
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int64_t OSSPlayer::GetCurrentPosition()
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{
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if (!playing)
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return 0;
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#ifdef SNDCTL_DSP_CURRENT_OPTR
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// OSS4
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long played_frames = 0;
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oss_count_t pos;
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if (ioctl(dspdev, SNDCTL_DSP_CURRENT_OPTR, &pos) >= 0) {
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// XXX: Apparently the semantics are different on FreeBSD...
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#ifdef __FREEBSD__
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played_frames = MAX(0, pos.samples - pos.fifo_samples);
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#else
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played_frames = pos.samples + pos.fifo_samples;
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#endif
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LOG_D("player/audio/oss") << "played_frames: " << played_frames << " fifo " << pos.fifo_samples;
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if (start_frame + played_frames >= end_frame) {
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if (displayTimer)
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displayTimer->Stop();
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}
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return start_frame + played_frames;
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}
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#endif
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// Fallback for old OSS versions
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int delay = 0;
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if (ioctl(dspdev, SNDCTL_DSP_GETODELAY, &delay) >= 0) {
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delay /= bpf;
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LOG_D("player/audio/oss") << "cur_frame: " << cur_frame << " delay " << delay;
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// delay can jitter a bit at the end, detect that
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if (cur_frame == end_frame && delay < rate / 20) {
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if (displayTimer)
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displayTimer->Stop();
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return cur_frame;
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}
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return MAX(0, (long) cur_frame - delay);
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}
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// Maybe this still didn't work...
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// Return the last written frame, timing will suffer
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return cur_frame;
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}
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/// @brief Thread constructor
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/// @param par
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///
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OSSPlayerThread::OSSPlayerThread(OSSPlayer *par) : wxThread(wxTHREAD_JOINABLE)
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{
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parent = par;
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}
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/// @brief Thread entry point
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/// @return
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///
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wxThread::ExitCode OSSPlayerThread::Entry() {
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// Use small enough writes for good timing accuracy with all
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// timing methods.
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const int wsize = parent->rate / 25;
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void *buf = malloc(wsize * parent->bpf);
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while (!TestDestroy() && parent->cur_frame < parent->end_frame) {
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int rsize = MIN(wsize, parent->end_frame - parent->cur_frame);
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parent->provider->GetAudioWithVolume(buf, parent->cur_frame,
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rsize, parent->volume);
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int written = ::write(parent->dspdev, buf, rsize * parent->bpf);
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parent->cur_frame += written / parent->bpf;
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}
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free(buf);
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parent->cur_frame = parent->end_frame;
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LOG_D("player/audio/oss") << "Thread dead";
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return 0;
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}
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#endif // WITH_OSS
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