forked from mia/Aegisub
df9c6b627f
Originally committed to SVN as r6223.
262 lines
7 KiB
C++
262 lines
7 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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// Update timer
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if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15);
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}
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void OpenALPlayer::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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// 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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if (timerToo && displayTimer) {
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displayTimer->Stop();
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}
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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(true);
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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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