931cc7f461
* Add LOG_(D|W|I)_IF for conditional logging. Originally committed to SVN as r4465.
381 lines
8.3 KiB
C++
381 lines
8.3 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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///////////
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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_openal.h"
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#include "audio_provider_manager.h"
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#include "frame_main.h"
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#include "options.h"
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#include "utils.h"
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#ifdef __WINDOWS__
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#include <al.h>
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#include <alc.h>
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#elif defined(__APPLE__)
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#include <OpenAL/AL.h>
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#include <OpenAL/ALC.h>
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#else
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#include <AL/al.h>
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#include <AL/alc.h>
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#endif
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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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/// @brief Constructor
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///
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OpenALPlayer::OpenALPlayer()
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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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}
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/// @brief Destructor
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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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/// @brief Open stream
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///
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void OpenALPlayer::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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device = alcOpenDevice(0);
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if (!device) {
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throw _T("Failed opening default OpenAL device");
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}
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// Create context
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context = alcCreateContext(device, 0);
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if (!context) {
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alcCloseDevice(device);
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throw _T("Failed creating OpenAL context");
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}
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if (!alcMakeContextCurrent(context)) {
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alcDestroyContext(context);
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alcCloseDevice(device);
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throw _T("Failed selecting OpenAL context");
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}
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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) {
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alcDestroyContext(context);
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alcCloseDevice(device);
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throw _T("Error generating OpenAL buffers");
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}
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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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alcDestroyContext(context);
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alcCloseDevice(device);
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throw _T("Error generating OpenAL source");
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}
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// Determine buffer length
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samplerate = provider->GetSampleRate();
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buffer_length = samplerate / 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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/// @brief Close stream
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/// @return
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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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// 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 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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/// @brief Stop
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/// @param timerToo
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/// @return
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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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/// @brief DOCME
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/// @param count
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///
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void OpenALPlayer::FillBuffers(ALsizei count)
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{
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LOG_D("player/audio/openal") << "count=" << count << " " << "buffers_free=" << buffers_free;
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if (count > buffers_free) count = buffers_free;
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if (count < 1) count = 1;
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// Get memory to hold sound buffers
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void *data = malloc(buffer_length * bpf);
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// Do the actual filling/queueing
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ALuint bufid = buf_first_free;
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while (count > 0) {
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ALsizei fill_len = buffer_length;
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if (fill_len > (ALsizei)(end_frame - cur_frame)) fill_len = (ALsizei)(end_frame - cur_frame);
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if (fill_len < 0) fill_len = 0;
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LOG_D("player/audio/openal") << "buffer_length=" << buffer_length << " fill_len=" << fill_len << " end_frame-cur-frame=" << 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(data, cur_frame, fill_len, volume);
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if (fill_len < buffer_length)
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// And zerofill the rest
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memset((char*)data+(fill_len*bpf), 0, (buffer_length-fill_len)*bpf);
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cur_frame += fill_len;
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alBufferData(buffers[bufid], AL_FORMAT_MONO16, data, buffer_length*bpf, samplerate);
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alSourceQueueBuffers(source, 1, &buffers[bufid]); // FIXME: collect buffer handles and queue all at once instead of one at a time?
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if (++bufid >= num_buffers) bufid = 0;
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count--; buffers_free--;
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}
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buf_first_free = bufid;
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// Free buffer memory again
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free(data);
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}
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/// @brief DOCME
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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 = new ALuint[newplayed];
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ALsizei i = 0;
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while (i < newplayed) {
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bufs[i++] = buffers[buf_first_queued];
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if (++buf_first_queued >= num_buffers) buf_first_queued = 0;
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}
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alSourceUnqueueBuffers(source, newplayed, bufs);
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delete[] 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) * buffer_length << " 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) * buffer_length > (ALsizei)(end_frame - start_frame)) {
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// Then stop
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Stop(true);
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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 OpenALPlayer::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 OpenALPlayer::SetEndPosition(int64_t pos)
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{
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end_frame = pos;
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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 OpenALPlayer::SetCurrentPosition(int64_t pos)
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{
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cur_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 OpenALPlayer::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 OpenALPlayer::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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///
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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 * buffer_length + start_frame + extra * samplerate / 1000;
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}
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#endif // WITH_OPENAL
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