/* * 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 "SlesOutputPlugin.hxx" #include "Object.hxx" #include "Engine.hxx" #include "Play.hxx" #include "AndroidSimpleBufferQueue.hxx" #include "../../OutputAPI.hxx" #include "../../Wrapper.hxx" #include "util/Macros.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" #include "system/ByteOrder.hxx" #include "Log.hxx" #include #include class SlesOutput { friend struct AudioOutputWrapper; static constexpr unsigned N_BUFFERS = 3; static constexpr size_t BUFFER_SIZE = 65536; AudioOutput base; SLES::Object engine_object, mix_object, play_object; SLES::Play play; SLES::AndroidSimpleBufferQueue queue; /** * This mutex protects the attributes "next" and "filled". It * is only needed while playback is launched, when the initial * buffers are being enqueued in the caller thread, while * another thread may invoke the registered callback. */ Mutex mutex; Cond cond; bool pause, cancel; /** * The number of buffers queued to OpenSLES. */ unsigned n_queued; /** * The index of the next buffer to be enqueued. */ unsigned next; /** * Does the "next" buffer already contain synthesised samples? * This can happen when PCMSynthesiser::Synthesise() has been * called, but the OpenSL/ES buffer queue was full. The * buffer will then be postponed. */ unsigned filled; /** * An array of buffers. It's one more than being managed by * OpenSL/ES, and the one not enqueued (see attribute #next) * will be written to. */ uint8_t buffers[N_BUFFERS][BUFFER_SIZE]; public: SlesOutput() :base(sles_output_plugin) {} operator AudioOutput *() { return &base; } bool Initialize(const config_param ¶m, Error &error) { return base.Configure(param, error); } bool Configure(const config_param ¶m, Error &error); bool Open(AudioFormat &audio_format, Error &error); void Close(); unsigned Delay() { return pause && !cancel ? 100 : 0; } size_t Play(const void *chunk, size_t size, Error &error); void Drain(); void Cancel(); bool Pause(); private: void PlayedCallback(); /** * OpenSL/ES callback which gets invoked when a buffer has * been consumed. It synthesises and enqueues the next * buffer. */ static void PlayedCallback(gcc_unused SLAndroidSimpleBufferQueueItf caller, void *pContext) { SlesOutput &sles = *(SlesOutput *)pContext; sles.PlayedCallback(); } }; static constexpr Domain sles_domain("sles"); inline bool SlesOutput::Configure(const config_param &, Error &) { return true; } inline bool SlesOutput::Open(AudioFormat &audio_format, Error &error) { SLresult result; SLObjectItf _object; result = slCreateEngine(&_object, 0, nullptr, 0, nullptr, nullptr); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "slCreateEngine() failed"); return false; } engine_object = SLES::Object(_object); result = engine_object.Realize(false); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Engine.Realize() failed"); engine_object.Destroy(); return false; } SLEngineItf _engine; result = engine_object.GetInterface(SL_IID_ENGINE, &_engine); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Engine.GetInterface(IID_ENGINE) failed"); engine_object.Destroy(); return false; } SLES::Engine engine(_engine); result = engine.CreateOutputMix(&_object, 0, nullptr, nullptr); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Engine.CreateOutputMix() failed"); engine_object.Destroy(); return false; } mix_object = SLES::Object(_object); result = mix_object.Realize(false); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Mix.Realize() failed"); mix_object.Destroy(); engine_object.Destroy(); return false; } SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, N_BUFFERS, }; if (audio_format.channels > 2) audio_format.channels = 1; SLDataFormat_PCM format_pcm; format_pcm.formatType = SL_DATAFORMAT_PCM; format_pcm.numChannels = audio_format.channels; /* from the Android NDK docs: "Note that the field samplesPerSec is actually in units of milliHz, despite the misleading name." */ format_pcm.samplesPerSec = audio_format.sample_rate * 1000u; format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; format_pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; format_pcm.channelMask = audio_format.channels == 1 ? SL_SPEAKER_FRONT_CENTER : SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; format_pcm.endianness = IsLittleEndian() ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; SLDataSource audioSrc = { &loc_bufq, &format_pcm }; SLDataLocator_OutputMix loc_outmix = { SL_DATALOCATOR_OUTPUTMIX, mix_object, }; SLDataSink audioSnk = { &loc_outmix, nullptr, }; const SLInterfaceID ids2[] = { SL_IID_PLAY, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION, }; static constexpr SLboolean req2[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, }; result = engine.CreateAudioPlayer(&_object, &audioSrc, &audioSnk, ARRAY_SIZE(ids2), ids2, req2); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Engine.CreateAudioPlayer() failed"); mix_object.Destroy(); engine_object.Destroy(); return false; } play_object = SLES::Object(_object); SLAndroidConfigurationItf android_config; if (play_object.GetInterface(SL_IID_ANDROIDCONFIGURATION, &android_config) == SL_RESULT_SUCCESS) { SLint32 stream_type = SL_ANDROID_STREAM_MEDIA; (*android_config)->SetConfiguration(android_config, SL_ANDROID_KEY_STREAM_TYPE, &stream_type, sizeof(stream_type)); } result = play_object.Realize(false); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.Realize() failed"); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); return false; } SLPlayItf _play; result = play_object.GetInterface(SL_IID_PLAY, &_play); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.GetInterface(IID_PLAY) failed"); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); return false; } play = SLES::Play(_play); SLAndroidSimpleBufferQueueItf _queue; result = play_object.GetInterface(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &_queue); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.GetInterface(IID_ANDROIDSIMPLEBUFFERQUEUE) failed"); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); return false; } queue = SLES::AndroidSimpleBufferQueue(_queue); result = queue.RegisterCallback(PlayedCallback, (void *)this); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.RegisterCallback() failed"); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); return false; } result = play.SetPlayState(SL_PLAYSTATE_PLAYING); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.SetPlayState(PLAYING) failed"); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); return false; } pause = cancel = false; n_queued = 0; next = 0; filled = 0; // TODO: support other sample formats audio_format.format = SampleFormat::S16; return true; } inline void SlesOutput::Close() { play.SetPlayState(SL_PLAYSTATE_STOPPED); play_object.Destroy(); mix_object.Destroy(); engine_object.Destroy(); } inline size_t SlesOutput::Play(const void *chunk, size_t size, Error &error) { cancel = false; if (pause) { SLresult result = play.SetPlayState(SL_PLAYSTATE_PLAYING); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "Play.SetPlayState(PLAYING) failed"); return false; } pause = false; } const ScopeLock protect(mutex); assert(filled < BUFFER_SIZE); while (n_queued == N_BUFFERS) { assert(filled == 0); cond.wait(mutex); } size_t nbytes = std::min(BUFFER_SIZE - filled, size); memcpy(buffers[next] + filled, chunk, nbytes); filled += nbytes; if (filled < BUFFER_SIZE) return nbytes; SLresult result = queue.Enqueue(buffers[next], BUFFER_SIZE); if (result != SL_RESULT_SUCCESS) { error.Set(sles_domain, int(result), "AndroidSimpleBufferQueue.Enqueue() failed"); return 0; } ++n_queued; next = (next + 1) % N_BUFFERS; filled = 0; return nbytes; } inline void SlesOutput::Drain() { const ScopeLock protect(mutex); assert(filled < BUFFER_SIZE); while (n_queued > 0) cond.wait(mutex); } inline void SlesOutput::Cancel() { pause = true; cancel = true; SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED); if (result != SL_RESULT_SUCCESS) FormatError(sles_domain, "Play.SetPlayState(PAUSED) failed"); result = queue.Clear(); if (result != SL_RESULT_SUCCESS) FormatWarning(sles_domain, "AndroidSimpleBufferQueue.Clear() failed"); const ScopeLock protect(mutex); n_queued = 0; filled = 0; } inline bool SlesOutput::Pause() { cancel = false; if (pause) return true; pause = true; SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED); if (result != SL_RESULT_SUCCESS) { FormatError(sles_domain, "Play.SetPlayState(PAUSED) failed"); return false; } return true; } inline void SlesOutput::PlayedCallback() { const ScopeLock protect(mutex); assert(n_queued > 0); --n_queued; cond.signal(); } static bool sles_test_default_device() { /* this is the default output plugin on Android, and it should be available in any case */ return true; } static AudioOutput * sles_output_init(const config_param ¶m, Error &error) { SlesOutput *sles = new SlesOutput(); if (!sles->Initialize(param, error) || !sles->Configure(param, error)) { delete sles; return nullptr; } return *sles; } typedef AudioOutputWrapper Wrapper; const struct AudioOutputPlugin sles_output_plugin = { "sles", sles_test_default_device, sles_output_init, &Wrapper::Finish, nullptr, nullptr, &Wrapper::Open, &Wrapper::Close, &Wrapper::Delay, nullptr, &Wrapper::Play, &Wrapper::Drain, &Wrapper::Cancel, &Wrapper::Pause, nullptr, };