ao: initial OpenSL ES support

OpenSL ES is used on Android. At the moment only stereo output is
supported. Two options are supported: 'frames-per-buffer' and
'sample-rate'. To get better latency the user of libmpv should pass
values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER)
and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).

audio/out/ao.c

@@ -43,6 +43,7 @@ extern const struct ao_driver audio_out_sndio;
 extern const struct ao_driver audio_out_pulse;
 extern const struct ao_driver audio_out_jack;
 extern const struct ao_driver audio_out_openal;
+extern const struct ao_driver audio_out_opensles;
 extern const struct ao_driver audio_out_null;
 extern const struct ao_driver audio_out_alsa;
 extern const struct ao_driver audio_out_wasapi;
@@ -74,6 +75,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_OPENAL
     &audio_out_openal,
 #endif
+#if HAVE_OPENSLES
+    &audio_out_opensles,
+#endif
 #if HAVE_SDL1 || HAVE_SDL2
     &audio_out_sdl,
 #endif

audio/out/ao_opensles.c

@@ -0,0 +1,250 @@
+/*
+ * OpenSL ES audio output driver.
+ * Copyright (C) 2016 Ilya Zhuravlev <whatever@xyz.is>
+ *
+ * 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 <http://www.gnu.org/licenses/>.
+ */
+
+#include "ao.h"
+#include "internal.h"
+#include "common/msg.h"
+#include "audio/format.h"
+#include "options/m_option.h"
+#include "osdep/timer.h"
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+
+#include <pthread.h>
+
+struct priv {
+    SLObjectItf sl, output_mix, player;
+    SLBufferQueueItf buffer_queue;
+    SLEngineItf engine;
+    SLPlayItf play;
+    char *curbuf, *buf1, *buf2;
+    size_t buffer_size;
+    pthread_mutex_t buffer_lock;
+
+    int cfg_frames_per_buffer;
+    int cfg_sample_rate;
+};
+
+static const int fmtmap[][2] = {
+    { AF_FORMAT_U8, SL_PCMSAMPLEFORMAT_FIXED_8 },
+    { AF_FORMAT_S16, SL_PCMSAMPLEFORMAT_FIXED_16 },
+    { AF_FORMAT_S32, SL_PCMSAMPLEFORMAT_FIXED_32 },
+    { 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->buf1);
+    free(p->buf2);
+    p->curbuf = p->buf1 = p->buf2 = 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->curbuf;
+    delay = 2 * p->buffer_size / (double)ao->bps;
+    ao_read_data(ao, data, p->buffer_size / ao->sstride,
+        mp_time_us() + 1000000LL * delay);
+
+    res = (*buffer_queue)->Enqueue(buffer_queue, p->curbuf, p->buffer_size);
+    if (res != SL_RESULT_SUCCESS)
+        MP_ERR(ao, "Failed to Enqueue: %d\n", res);
+    else
+        p->curbuf = (p->curbuf == p->buf1) ? p->buf2 : p->buf1;
+
+    pthread_mutex_unlock(&p->buffer_lock);
+}
+
+#define DEFAULT_BUFFER_SIZE_MS 50
+
+#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 = 2;
+
+    pcm.formatType = SL_DATAFORMAT_PCM;
+    pcm.numChannels = 2;
+
+    int compatible_formats[AF_FORMAT_COUNT];
+    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;
+
+    if (p->cfg_sample_rate)
+        ao->samplerate = p->cfg_sample_rate;
+
+    // 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->buf1 = calloc(1, p->buffer_size);
+    p->buf2 = calloc(1, p->buffer_size);
+    p->curbuf = p->buf1;
+    if (!p->buf1 || !p->buf2) {
+        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;
+
+    SLboolean required[] = { SL_BOOLEAN_TRUE };
+    SLInterfaceID iid_array[] = { SL_IID_BUFFERQUEUE };
+    CHK((*p->engine)->CreateAudioPlayer(p->engine, &p->player, &audio_source,
+        &audio_sink, 1, 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));
+
+    return 1;
+error:
+    uninit(ao);
+    return -1;
+}
+
+#undef CHK
+
+static void set_play_state(struct ao *ao, SLuint32 state)
+{
+    struct priv *p = ao->priv;
+    SLresult res = (*p->play)->SetPlayState(p->play, state);
+    if (res != SL_RESULT_SUCCESS)
+        MP_ERR(ao, "Failed to SetPlayState(%d): %d\n", state, res);
+}
+
+static void reset(struct ao *ao)
+{
+    set_play_state(ao, SL_PLAYSTATE_STOPPED);
+}
+
+static void resume(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    set_play_state(ao, SL_PLAYSTATE_PLAYING);
+
+    // enqueue two buffers
+    buffer_callback(p->buffer_queue, ao);
+    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, 10000),
+        OPT_INTRANGE("sample-rate", cfg_sample_rate, 0, 1000, 100000),
+        {0}
+    },
+};

wscript

@@ -555,6 +555,10 @@ audio_output_features = [
         'desc': 'OpenAL audio output',
         'func': check_pkg_config('openal', '>= 1.13'),
         'default': 'disable'
+    }, {
+        'name': '--opensles',
+        'desc': 'OpenSL ES audio output',
+        'func': check_statement('SLES/OpenSLES.h', 'slCreateEngine', lib="OpenSLES"),
     }, {
         'name': '--alsa',
         'desc': 'ALSA audio output',

wscript_build.py

@@ -138,6 +138,7 @@ def build(ctx):
         ( "audio/out/ao_lavc.c",                 "encoding" ),
         ( "audio/out/ao_null.c" ),
         ( "audio/out/ao_openal.c",               "openal" ),
+        ( "audio/out/ao_opensles.c",             "opensles" ),
         ( "audio/out/ao_oss.c",                  "oss-audio" ),
         ( "audio/out/ao_pcm.c" ),
         ( "audio/out/ao_pulse.c",                "pulse" ),