/* * Copyright (C) 2003-2015 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with this program; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include "config.h" #include "OpenALOutputPlugin.hxx" #include "../OutputAPI.hxx" #include "../Wrapper.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" #include #ifndef __APPLE__ #include #include #else #include #include #endif /* should be enough for buffer size = 2048 */ #define NUM_BUFFERS 16 struct OpenALOutput { friend struct AudioOutputWrapper; AudioOutput base; const char *device_name; ALCdevice *device; ALCcontext *context; ALuint buffers[NUM_BUFFERS]; unsigned filled; ALuint source; ALenum format; ALuint frequency; OpenALOutput() :base(openal_output_plugin) {} bool Configure(const config_param ¶m, Error &error); static OpenALOutput *Create(const config_param ¶m, Error &error); bool Open(AudioFormat &audio_format, Error &error); void Close(); gcc_pure unsigned Delay() const { return filled < NUM_BUFFERS || HasProcessed() ? 0 /* we don't know exactly how long we must wait for the next buffer to finish, so this is a random guess: */ : 50; } size_t Play(const void *chunk, size_t size, Error &error); void Cancel(); private: gcc_pure ALint GetSourceI(ALenum param) const { ALint value; alGetSourcei(source, param, &value); return value; } gcc_pure bool HasProcessed() const { return GetSourceI(AL_BUFFERS_PROCESSED) > 0; } gcc_pure bool IsPlaying() const { return GetSourceI(AL_SOURCE_STATE) == AL_PLAYING; } bool SetupContext(Error &error); }; static constexpr Domain openal_output_domain("openal_output"); static ALenum openal_audio_format(AudioFormat &audio_format) { /* note: cannot map SampleFormat::S8 to AL_FORMAT_STEREO8 or AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit samples, while MPD uses signed samples */ switch (audio_format.format) { case SampleFormat::S16: if (audio_format.channels == 2) return AL_FORMAT_STEREO16; if (audio_format.channels == 1) return AL_FORMAT_MONO16; /* fall back to mono */ audio_format.channels = 1; return openal_audio_format(audio_format); default: /* fall back to 16 bit */ audio_format.format = SampleFormat::S16; return openal_audio_format(audio_format); } } inline bool OpenALOutput::SetupContext(Error &error) { device = alcOpenDevice(device_name); if (device == nullptr) { error.Format(openal_output_domain, "Error opening OpenAL device \"%s\"", device_name); return false; } context = alcCreateContext(device, nullptr); if (context == nullptr) { error.Format(openal_output_domain, "Error creating context for \"%s\"", device_name); alcCloseDevice(device); return false; } return true; } inline bool OpenALOutput::Configure(const config_param ¶m, Error &error) { if (!base.Configure(param, error)) return false; device_name = param.GetBlockValue("device"); if (device_name == nullptr) device_name = alcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER); return true; } inline OpenALOutput * OpenALOutput::Create(const config_param ¶m, Error &error) { OpenALOutput *oo = new OpenALOutput(); if (!oo->Configure(param, error)) { delete oo; return nullptr; } return oo; } inline bool OpenALOutput::Open(AudioFormat &audio_format, Error &error) { format = openal_audio_format(audio_format); if (!SetupContext(error)) return false; alcMakeContextCurrent(context); alGenBuffers(NUM_BUFFERS, buffers); if (alGetError() != AL_NO_ERROR) { error.Set(openal_output_domain, "Failed to generate buffers"); return false; } alGenSources(1, &source); if (alGetError() != AL_NO_ERROR) { error.Set(openal_output_domain, "Failed to generate source"); alDeleteBuffers(NUM_BUFFERS, buffers); return false; } filled = 0; frequency = audio_format.sample_rate; return true; } inline void OpenALOutput::Close() { alcMakeContextCurrent(context); alDeleteSources(1, &source); alDeleteBuffers(NUM_BUFFERS, buffers); alcDestroyContext(context); alcCloseDevice(device); } inline size_t OpenALOutput::Play(const void *chunk, size_t size, gcc_unused Error &error) { if (alcGetCurrentContext() != context) alcMakeContextCurrent(context); ALuint buffer; if (filled < NUM_BUFFERS) { /* fill all buffers */ buffer = buffers[filled]; filled++; } else { /* wait for processed buffer */ while (!HasProcessed()) usleep(10); alSourceUnqueueBuffers(source, 1, &buffer); } alBufferData(buffer, format, chunk, size, frequency); alSourceQueueBuffers(source, 1, &buffer); if (!IsPlaying()) alSourcePlay(source); return size; } inline void OpenALOutput::Cancel() { filled = 0; alcMakeContextCurrent(context); alSourceStop(source); /* force-unqueue all buffers */ alSourcei(source, AL_BUFFER, 0); filled = 0; } typedef AudioOutputWrapper Wrapper; const struct AudioOutputPlugin openal_output_plugin = { "openal", nullptr, &Wrapper::Init, &Wrapper::Finish, nullptr, nullptr, &Wrapper::Open, &Wrapper::Close, &Wrapper::Delay, nullptr, &Wrapper::Play, nullptr, &Wrapper::Cancel, nullptr, nullptr, };