// Copyright (c) 2007, Niels Martin Hansen // All rights reserved. // // 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 // // Website: http://aegisub.cellosoft.com // Contact: mailto:jiifurusu@gmail.com // #ifdef WITH_OPENAL /////////// // Headers #include #include "audio_player.h" #include "audio_provider_manager.h" #include "utils.h" #include "main.h" #include "frame_main.h" #include "audio_player_openal.h" #include "options.h" #ifdef __WINDOWS__ #include #include #elif defined(__APPLE__) #include #include #else #include #include #endif // Auto-link to OpenAL lib for MSVC #ifdef _MSC_VER #pragma comment(lib, "openal32.lib") #endif /////////////// // Constructor OpenALPlayer::OpenALPlayer() { volume = 1.0f; open = false; playing = false; start_frame = cur_frame = end_frame = bpf = 0; provider = 0; } ////////////// // Destructor OpenALPlayer::~OpenALPlayer() { CloseStream(); } /////////////// // Open stream void OpenALPlayer::OpenStream() { CloseStream(); // Get provider provider = GetProvider(); bpf = provider->GetChannels() * provider->GetBytesPerSample(); // Open device device = alcOpenDevice(0); if (!device) { throw _T("Failed opening default OpenAL device"); } // Create context context = alcCreateContext(device, 0); if (!context) { alcCloseDevice(device); throw _T("Failed creating OpenAL context"); } if (!alcMakeContextCurrent(context)) { alcDestroyContext(context); alcCloseDevice(device); throw _T("Failed selecting OpenAL context"); } // Clear error code alGetError(); // Generate buffers alGenBuffers(num_buffers, buffers); if (alGetError() != AL_NO_ERROR) { alcDestroyContext(context); alcCloseDevice(device); throw _T("Error generating OpenAL buffers"); } // Generate source alGenSources(1, &source); if (alGetError() != AL_NO_ERROR) { alDeleteBuffers(num_buffers, buffers); alcDestroyContext(context); alcCloseDevice(device); throw _T("Error generating OpenAL source"); } // Determine buffer length samplerate = provider->GetSampleRate(); buffer_length = samplerate / num_buffers / 2; // buffers for half a second of audio // Now ready open = true; } //////////////// // Close stream void OpenALPlayer::CloseStream() { if (!open) return; Stop(); alDeleteSources(1, &source); alDeleteBuffers(num_buffers, buffers); alcDestroyContext(context); alcCloseDevice(device); // No longer working open = false; } //////// // Play void OpenALPlayer::Play(int64_t start,int64_t count) { if (playing) { // Quick reset playing = false; alSourceStop(source); alSourcei(source, AL_BUFFER, 0); } // Set params start_frame = start; cur_frame = start; end_frame = start + count; playing = true; // Prepare buffers buffers_free = num_buffers; buffers_played = 0; buf_first_free = 0; buf_first_queued = 0; FillBuffers(num_buffers); // And go! alSourcePlay(source); wxTimer::Start(100); playback_segment_timer.Start(); // Update timer if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15); } //////// // Stop void OpenALPlayer::Stop(bool timerToo) { if (!open) return; if (!playing) return; // Reset data wxTimer::Stop(); playing = false; start_frame = 0; cur_frame = 0; end_frame = 0; // Then drop the playback alSourceStop(source); alSourcei(source, AL_BUFFER, 0); if (timerToo && displayTimer) { displayTimer->Stop(); } } void OpenALPlayer::FillBuffers(ALsizei count) { wxLogDebug(_T("FillBuffers: count=%d, buffers_free=%d"), count, buffers_free); if (count > buffers_free) count = buffers_free; if (count < 1) count = 1; // Get memory to hold sound buffers void *data = malloc(buffer_length * bpf); // Do the actual filling/queueing ALuint bufid = buf_first_free; while (count > 0) { ALsizei fill_len = buffer_length; if (fill_len > end_frame - cur_frame) fill_len = end_frame - cur_frame; wxLogDebug(_T("buffer_length=%d, fill_len=%d, end_frame-cur_frame=%d"), buffer_length, fill_len, end_frame-cur_frame); if (fill_len > 0) // Get fill_len frames of audio provider->GetAudioWithVolume(data, cur_frame, fill_len, volume); if (fill_len < buffer_length) // And zerofill the rest memset((char*)data+(fill_len*bpf), 0, (buffer_length-fill_len)*bpf); cur_frame += fill_len; alBufferData(buffers[bufid], AL_FORMAT_MONO16, data, buffer_length*bpf, samplerate); alSourceQueueBuffers(source, 1, &buffers[bufid]); // FIXME: collect buffer handles and queue all at once instead of one at a time? if (++bufid >= num_buffers) bufid = 0; count--; buffers_free--; } buf_first_free = bufid; // Free buffer memory again free(data); } void OpenALPlayer::Notify() { ALsizei newplayed; alGetSourcei(source, AL_BUFFERS_PROCESSED, &newplayed); wxLogDebug(_T("OpenAL Player notify: buffers_played=%d, newplayed=%d, playeddiff=%d"), buffers_played, newplayed); if (newplayed > 0) { // Reclaim buffers ALuint *bufs = new ALuint[newplayed]; ALsizei i = 0; while (i < newplayed) { bufs[i++] = buffers[buf_first_queued]; if (++buf_first_queued >= num_buffers) buf_first_queued = 0; } alSourceUnqueueBuffers(source, newplayed, bufs); delete[] bufs; buffers_free += newplayed; // Update buffers_played += newplayed; playback_segment_timer.Start(); // Fill more buffers FillBuffers(newplayed); } wxLogDebug(_T("frames played=%d, num frames=%d"), (buffers_played - num_buffers) * buffer_length, end_frame - start_frame); // Check that all of the selected audio plus one full set of buffers has been queued if ((buffers_played - num_buffers) * buffer_length > (ALsizei)(end_frame - start_frame)) { // Then stop Stop(true); } } bool OpenALPlayer::IsPlaying() { return playing; } /////////// // Set end void OpenALPlayer::SetEndPosition(int64_t pos) { end_frame = pos; } //////////////////////// // Set current position void OpenALPlayer::SetCurrentPosition(int64_t pos) { cur_frame = pos; } int64_t OpenALPlayer::GetStartPosition() { return start_frame; } int64_t OpenALPlayer::GetEndPosition() { return end_frame; } //////////////////////// // Get current position int64_t OpenALPlayer::GetCurrentPosition() { // FIXME: this should be based on not duration played but actual sample being heard // (during vidoeo playback, cur_frame might get changed to resync) long extra = playback_segment_timer.Time(); return buffers_played * buffer_length + start_frame + extra * samplerate / 1000; } #endif // WITH_OPENAL