forked from mia/Aegisub
7218c04d52
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.
255 lines
6.8 KiB
C++
255 lines
6.8 KiB
C++
// Copyright (c) 2007, Niels Martin Hansen
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// All rights reserved.
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//
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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_openal.cpp
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/// @brief OpenAL-based 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_OPENAL
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#include <libaegisub/log.h>
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#include "audio_player_openal.h"
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#include "audio_controller.h"
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#include "utils.h"
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// Auto-link to OpenAL lib for MSVC
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#ifdef _MSC_VER
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#pragma comment(lib, "openal32.lib")
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#endif
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DEFINE_SIMPLE_EXCEPTION(OpenALException, agi::AudioPlayerOpenError, "audio/open/player/openal")
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OpenALPlayer::OpenALPlayer()
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: open(false)
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, playing(false)
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, volume(1.f)
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, samplerate(0)
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, bpf(0)
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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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, device(0)
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, context(0)
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{
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}
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OpenALPlayer::~OpenALPlayer()
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{
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CloseStream();
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}
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void OpenALPlayer::OpenStream()
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{
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CloseStream();
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bpf = provider->GetChannels() * provider->GetBytesPerSample();
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try {
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// Open device
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device = alcOpenDevice(0);
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if (!device) throw OpenALException("Failed opening default OpenAL device", 0);
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// Create context
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context = alcCreateContext(device, 0);
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if (!context) throw OpenALException("Failed creating OpenAL context", 0);
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if (!alcMakeContextCurrent(context)) throw OpenALException("Failed selecting OpenAL context", 0);
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// Clear error code
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alGetError();
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// Generate buffers
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alGenBuffers(num_buffers, buffers);
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if (alGetError() != AL_NO_ERROR) throw OpenALException("Error generating OpenAL buffers", 0);
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// Generate source
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alGenSources(1, &source);
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if (alGetError() != AL_NO_ERROR) {
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alDeleteBuffers(num_buffers, buffers);
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throw OpenALException("Error generating OpenAL source", 0);
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}
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}
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catch (...)
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{
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alcDestroyContext(context);
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alcCloseDevice(device);
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context = 0;
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device = 0;
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throw;
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}
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// Determine buffer length
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samplerate = provider->GetSampleRate();
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decode_buffer.resize(samplerate * bpf / num_buffers / 2); // buffers for half a second of audio
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// Now ready
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open = true;
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}
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void OpenALPlayer::CloseStream()
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{
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if (!open) return;
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Stop();
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alDeleteSources(1, &source);
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alDeleteBuffers(num_buffers, buffers);
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alcDestroyContext(context);
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alcCloseDevice(device);
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context = 0;
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device = 0;
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// No longer working
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open = false;
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}
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void OpenALPlayer::Play(int64_t start, int64_t count)
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{
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if (playing) {
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// Quick reset
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playing = false;
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alSourceStop(source);
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alSourcei(source, AL_BUFFER, 0);
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}
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// Set params
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start_frame = start;
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cur_frame = start;
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end_frame = start + count;
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playing = true;
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// Prepare buffers
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buffers_free = num_buffers;
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buffers_played = 0;
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buf_first_free = 0;
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buf_first_queued = 0;
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FillBuffers(num_buffers);
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// And go!
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alSourcePlay(source);
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wxTimer::Start(100);
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playback_segment_timer.Start();
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}
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void OpenALPlayer::Stop()
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{
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if (!open) return;
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if (!playing) return;
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// Reset data
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wxTimer::Stop();
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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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// Then drop the playback
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alSourceStop(source);
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alSourcei(source, AL_BUFFER, 0);
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}
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void OpenALPlayer::FillBuffers(ALsizei count)
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{
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// Do the actual filling/queueing
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for (count = mid(1, count, buffers_free); count > 0; --count) {
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ALsizei fill_len = mid<ALsizei>(0, decode_buffer.size() / bpf, end_frame - cur_frame);
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if (fill_len > 0)
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// Get fill_len frames of audio
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provider->GetAudioWithVolume(&decode_buffer[0], cur_frame, fill_len, volume);
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if ((size_t)fill_len * bpf < decode_buffer.size())
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// And zerofill the rest
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memset(&decode_buffer[fill_len * bpf], 0, decode_buffer.size() - fill_len * bpf);
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cur_frame += fill_len;
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alBufferData(buffers[buf_first_free], AL_FORMAT_MONO16, &decode_buffer[0], decode_buffer.size(), samplerate);
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alSourceQueueBuffers(source, 1, &buffers[buf_first_free]); // FIXME: collect buffer handles and queue all at once instead of one at a time?
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buf_first_free = (buf_first_free + 1) % num_buffers;
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--buffers_free;
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}
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}
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void OpenALPlayer::Notify()
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{
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ALsizei newplayed;
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alGetSourcei(source, AL_BUFFERS_PROCESSED, &newplayed);
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LOG_D("player/audio/openal") << "buffers_played=" << buffers_played << " newplayed=" << newplayed;
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if (newplayed > 0) {
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// Reclaim buffers
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ALuint bufs[num_buffers];
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for (ALsizei i = 0; i < newplayed; ++i) {
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bufs[i] = buffers[buf_first_queued];
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buf_first_queued = (buf_first_queued + 1) % num_buffers;
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}
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alSourceUnqueueBuffers(source, newplayed, bufs);
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buffers_free += newplayed;
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// Update
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buffers_played += newplayed;
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playback_segment_timer.Start();
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// Fill more buffers
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FillBuffers(newplayed);
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}
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LOG_D("player/audio/openal") << "frames played=" << (buffers_played - num_buffers) * decode_buffer.size() / bpf << " num frames=" << end_frame - start_frame;
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// Check that all of the selected audio plus one full set of buffers has been queued
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if ((buffers_played - num_buffers) * (int64_t)decode_buffer.size() > (end_frame - start_frame) * bpf) {
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Stop();
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}
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}
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void OpenALPlayer::SetEndPosition(int64_t pos)
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{
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end_frame = pos;
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}
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void OpenALPlayer::SetCurrentPosition(int64_t pos)
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{
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cur_frame = pos;
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}
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int64_t OpenALPlayer::GetCurrentPosition()
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{
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// FIXME: this should be based on not duration played but actual sample being heard
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// (during video playback, cur_frame might get changed to resync)
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long extra = playback_segment_timer.Time();
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return buffers_played * decode_buffer.size() / bpf + start_frame + extra * samplerate / 1000;
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}
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#endif // WITH_OPENAL
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