/* * OpenSL ES audio output driver. * Copyright (C) 2016 Ilya Zhuravlev * * This file is part of mpv. * * mpv is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * mpv 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 Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with mpv. If not, see . */ #include "ao.h" #include "internal.h" #include "common/msg.h" #include "audio/format.h" #include "options/m_option.h" #include "osdep/timer.h" #include #include #include struct priv { SLObjectItf sl, output_mix, player; SLBufferQueueItf buffer_queue; SLEngineItf engine; SLPlayItf play; char *buf; size_t buffer_size; pthread_mutex_t buffer_lock; double audio_latency; int cfg_frames_per_buffer; }; static const int fmtmap[][2] = { { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 }, { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 }, { 0 } }; #define DESTROY(thing) \ if (p->thing) { \ (*p->thing)->Destroy(p->thing); \ p->thing = NULL; \ } static void uninit(struct ao *ao) { struct priv *p = ao->priv; DESTROY(player); DESTROY(output_mix); DESTROY(sl); p->buffer_queue = NULL; p->engine = NULL; p->play = NULL; pthread_mutex_destroy(&p->buffer_lock); free(p->buf); p->buf = NULL; p->buffer_size = 0; } #undef DESTROY static void buffer_callback(SLBufferQueueItf buffer_queue, void *context) { struct ao *ao = context; struct priv *p = ao->priv; SLresult res; void *data[1]; double delay; pthread_mutex_lock(&p->buffer_lock); data[0] = p->buf; delay = 2 * p->buffer_size / (double)ao->bps; delay += p->audio_latency; ao_read_data(ao, data, p->buffer_size / ao->sstride, mp_time_us() + 1000000LL * delay); res = (*buffer_queue)->Enqueue(buffer_queue, p->buf, p->buffer_size); if (res != SL_RESULT_SUCCESS) MP_ERR(ao, "Failed to Enqueue: %d\n", res); pthread_mutex_unlock(&p->buffer_lock); } #define DEFAULT_BUFFER_SIZE_MS 250 #define CHK(stmt) \ { \ SLresult res = stmt; \ if (res != SL_RESULT_SUCCESS) { \ MP_ERR(ao, "%s: %d\n", #stmt, res); \ goto error; \ } \ } static int init(struct ao *ao) { struct priv *p = ao->priv; SLDataLocator_BufferQueue locator_buffer_queue; SLDataLocator_OutputMix locator_output_mix; SLDataFormat_PCM pcm; SLDataSource audio_source; SLDataSink audio_sink; // This AO only supports two channels at the moment mp_chmap_from_channels(&ao->channels, 2); CHK(slCreateEngine(&p->sl, 0, NULL, 0, NULL, NULL)); CHK((*p->sl)->Realize(p->sl, SL_BOOLEAN_FALSE)); CHK((*p->sl)->GetInterface(p->sl, SL_IID_ENGINE, (void*)&p->engine)); CHK((*p->engine)->CreateOutputMix(p->engine, &p->output_mix, 0, NULL, NULL)); CHK((*p->output_mix)->Realize(p->output_mix, SL_BOOLEAN_FALSE)); locator_buffer_queue.locatorType = SL_DATALOCATOR_BUFFERQUEUE; locator_buffer_queue.numBuffers = 1; pcm.formatType = SL_DATAFORMAT_PCM; pcm.numChannels = 2; int compatible_formats[AF_FORMAT_COUNT + 1]; af_get_best_sample_formats(ao->format, compatible_formats); pcm.bitsPerSample = 0; for (int i = 0; compatible_formats[i] && !pcm.bitsPerSample; ++i) for (int j = 0; fmtmap[j][0]; ++j) if (compatible_formats[i] == fmtmap[j][0]) { ao->format = fmtmap[j][0]; pcm.bitsPerSample = fmtmap[j][1]; break; } if (!pcm.bitsPerSample) { MP_ERR(ao, "Cannot find compatible audio format\n"); goto error; } pcm.containerSize = 8 * af_fmt_to_bytes(ao->format); pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; // samplesPerSec is misnamed, actually it's samples per ms pcm.samplesPerSec = ao->samplerate * 1000; if (p->cfg_frames_per_buffer) ao->device_buffer = p->cfg_frames_per_buffer; else ao->device_buffer = ao->samplerate * DEFAULT_BUFFER_SIZE_MS / 1000; p->buffer_size = ao->device_buffer * ao->channels.num * af_fmt_to_bytes(ao->format); p->buf = calloc(1, p->buffer_size); if (!p->buf) { MP_ERR(ao, "Failed to allocate device buffer\n"); goto error; } int r = pthread_mutex_init(&p->buffer_lock, NULL); if (r) { MP_ERR(ao, "Failed to initialize the mutex: %d\n", r); goto error; } audio_source.pFormat = (void*)&pcm; audio_source.pLocator = (void*)&locator_buffer_queue; locator_output_mix.locatorType = SL_DATALOCATOR_OUTPUTMIX; locator_output_mix.outputMix = p->output_mix; audio_sink.pLocator = (void*)&locator_output_mix; audio_sink.pFormat = NULL; SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION }; SLboolean required[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE }; CHK((*p->engine)->CreateAudioPlayer(p->engine, &p->player, &audio_source, &audio_sink, 2, iid_array, required)); CHK((*p->player)->Realize(p->player, SL_BOOLEAN_FALSE)); CHK((*p->player)->GetInterface(p->player, SL_IID_PLAY, (void*)&p->play)); CHK((*p->player)->GetInterface(p->player, SL_IID_BUFFERQUEUE, (void*)&p->buffer_queue)); CHK((*p->buffer_queue)->RegisterCallback(p->buffer_queue, buffer_callback, ao)); CHK((*p->play)->SetPlayState(p->play, SL_PLAYSTATE_PLAYING)); SLAndroidConfigurationItf android_config; SLuint32 audio_latency = 0, value_size = sizeof(SLuint32); SLint32 get_interface_result = (*p->player)->GetInterface( p->player, SL_IID_ANDROIDCONFIGURATION, &android_config ); if (get_interface_result == SL_RESULT_SUCCESS) { SLint32 get_configuration_result = (*android_config)->GetConfiguration( android_config, (const SLchar *)"androidGetAudioLatency", &value_size, &audio_latency ); if (get_configuration_result == SL_RESULT_SUCCESS) { p->audio_latency = (double)audio_latency / 1000.0; MP_INFO(ao, "Device latency is %f\n", p->audio_latency); } } return 1; error: uninit(ao); return -1; } #undef CHK static void reset(struct ao *ao) { struct priv *p = ao->priv; (*p->buffer_queue)->Clear(p->buffer_queue); } static void resume(struct ao *ao) { struct priv *p = ao->priv; buffer_callback(p->buffer_queue, ao); } #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_opensles = { .description = "OpenSL ES audio output", .name = "opensles", .init = init, .uninit = uninit, .reset = reset, .resume = resume, .priv_size = sizeof(struct priv), .options = (const struct m_option[]) { OPT_INTRANGE("frames-per-buffer", cfg_frames_per_buffer, 0, 1, 96000), {0} }, .options_prefix = "opensles", };