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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/libcubeb/src/cubeb_opensl.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/libcubeb/src/cubeb_opensl.c')
-rw-r--r--media/libcubeb/src/cubeb_opensl.c889
1 files changed, 889 insertions, 0 deletions
diff --git a/media/libcubeb/src/cubeb_opensl.c b/media/libcubeb/src/cubeb_opensl.c
new file mode 100644
index 000000000..d44a56bd7
--- /dev/null
+++ b/media/libcubeb/src/cubeb_opensl.c
@@ -0,0 +1,889 @@
+/*
+ * Copyright © 2012 Mozilla Foundation
+ *
+ * This program is made available under an ISC-style license. See the
+ * accompanying file LICENSE for details.
+ */
+#undef NDEBUG
+#include <assert.h>
+#include <dlfcn.h>
+#include <stdlib.h>
+#include <pthread.h>
+#include <SLES/OpenSLES.h>
+#include <math.h>
+#include <time.h>
+#if defined(__ANDROID__)
+#include <dlfcn.h>
+#include <sys/system_properties.h>
+#include "android/sles_definitions.h"
+#include <SLES/OpenSLES_Android.h>
+#include <android/log.h>
+#include <android/api-level.h>
+#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "Cubeb_OpenSL" , ## args)
+#define ANDROID_VERSION_GINGERBREAD_MR1 10
+#define ANDROID_VERSION_LOLLIPOP 21
+#define ANDROID_VERSION_MARSHMALLOW 23
+#endif
+#include "cubeb/cubeb.h"
+#include "cubeb-internal.h"
+#include "cubeb_resampler.h"
+#include "cubeb-sles.h"
+
+static struct cubeb_ops const opensl_ops;
+
+struct cubeb {
+ struct cubeb_ops const * ops;
+ void * lib;
+ void * libmedia;
+ int32_t (* get_output_latency)(uint32_t * latency, int stream_type);
+ SLInterfaceID SL_IID_BUFFERQUEUE;
+ SLInterfaceID SL_IID_PLAY;
+#if defined(__ANDROID__)
+ SLInterfaceID SL_IID_ANDROIDCONFIGURATION;
+#endif
+ SLInterfaceID SL_IID_VOLUME;
+ SLObjectItf engObj;
+ SLEngineItf eng;
+ SLObjectItf outmixObj;
+};
+
+#define NELEMS(A) (sizeof(A) / sizeof A[0])
+#define NBUFS 4
+#define AUDIO_STREAM_TYPE_MUSIC 3
+
+struct cubeb_stream {
+ cubeb * context;
+ pthread_mutex_t mutex;
+ SLObjectItf playerObj;
+ SLPlayItf play;
+ SLBufferQueueItf bufq;
+ SLVolumeItf volume;
+ uint8_t *queuebuf[NBUFS];
+ int queuebuf_idx;
+ long queuebuf_len;
+ long bytespersec;
+ long framesize;
+ long written;
+ int draining;
+ cubeb_stream_type stream_type;
+
+ cubeb_data_callback data_callback;
+ cubeb_state_callback state_callback;
+ void * user_ptr;
+
+ cubeb_resampler * resampler;
+ unsigned int inputrate;
+ unsigned int outputrate;
+ unsigned int latency;
+ int64_t lastPosition;
+ int64_t lastPositionTimeStamp;
+ int64_t lastCompensativePosition;
+};
+
+static void
+play_callback(SLPlayItf caller, void * user_ptr, SLuint32 event)
+{
+ cubeb_stream * stm = user_ptr;
+ int draining;
+ assert(stm);
+ switch (event) {
+ case SL_PLAYEVENT_HEADATMARKER:
+ pthread_mutex_lock(&stm->mutex);
+ draining = stm->draining;
+ pthread_mutex_unlock(&stm->mutex);
+ if (draining) {
+ stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
+ (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
+ }
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+bufferqueue_callback(SLBufferQueueItf caller, void * user_ptr)
+{
+ cubeb_stream * stm = user_ptr;
+ assert(stm);
+ SLBufferQueueState state;
+ SLresult res;
+
+ res = (*stm->bufq)->GetState(stm->bufq, &state);
+ assert(res == SL_RESULT_SUCCESS);
+
+ if (state.count > 1)
+ return;
+
+ SLuint32 i;
+ for (i = state.count; i < NBUFS; i++) {
+ uint8_t *buf = stm->queuebuf[stm->queuebuf_idx];
+ long written = 0;
+ pthread_mutex_lock(&stm->mutex);
+ int draining = stm->draining;
+ pthread_mutex_unlock(&stm->mutex);
+
+ if (!draining) {
+ written = cubeb_resampler_fill(stm->resampler,
+ NULL, NULL,
+ buf, stm->queuebuf_len / stm->framesize);
+ if (written < 0 || written * stm->framesize > stm->queuebuf_len) {
+ (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
+ return;
+ }
+ }
+
+ // Keep sending silent data even in draining mode to prevent the audio
+ // back-end from being stopped automatically by OpenSL/ES.
+ memset(buf + written * stm->framesize, 0, stm->queuebuf_len - written * stm->framesize);
+ res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->queuebuf_len);
+ assert(res == SL_RESULT_SUCCESS);
+ stm->queuebuf_idx = (stm->queuebuf_idx + 1) % NBUFS;
+ if (written > 0) {
+ pthread_mutex_lock(&stm->mutex);
+ stm->written += written;
+ pthread_mutex_unlock(&stm->mutex);
+ }
+
+ if (!draining && written * stm->framesize < stm->queuebuf_len) {
+ pthread_mutex_lock(&stm->mutex);
+ int64_t written_duration = INT64_C(1000) * stm->written * stm->framesize / stm->bytespersec;
+ stm->draining = 1;
+ pthread_mutex_unlock(&stm->mutex);
+ // Use SL_PLAYEVENT_HEADATMARKER event from slPlayCallback of SLPlayItf
+ // to make sure all the data has been processed.
+ (*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)written_duration);
+ return;
+ }
+ }
+}
+
+#if defined(__ANDROID__)
+static SLuint32
+convert_stream_type_to_sl_stream(cubeb_stream_type stream_type)
+{
+ switch(stream_type) {
+ case CUBEB_STREAM_TYPE_SYSTEM:
+ return SL_ANDROID_STREAM_SYSTEM;
+ case CUBEB_STREAM_TYPE_MUSIC:
+ return SL_ANDROID_STREAM_MEDIA;
+ case CUBEB_STREAM_TYPE_NOTIFICATION:
+ return SL_ANDROID_STREAM_NOTIFICATION;
+ case CUBEB_STREAM_TYPE_ALARM:
+ return SL_ANDROID_STREAM_ALARM;
+ case CUBEB_STREAM_TYPE_VOICE_CALL:
+ return SL_ANDROID_STREAM_VOICE;
+ case CUBEB_STREAM_TYPE_RING:
+ return SL_ANDROID_STREAM_RING;
+ case CUBEB_STREAM_TYPE_SYSTEM_ENFORCED:
+ return SL_ANDROID_STREAM_SYSTEM_ENFORCED;
+ default:
+ return 0xFFFFFFFF;
+ }
+}
+#endif
+
+static void opensl_destroy(cubeb * ctx);
+
+#if defined(__ANDROID__)
+
+// The bionic header file on B2G contains the required
+// declarations on all releases.
+#ifndef MOZ_WIDGET_GONK
+
+#if (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP)
+typedef int (system_property_get)(const char*, char*);
+
+static int
+__system_property_get(const char* name, char* value)
+{
+ void* libc = dlopen("libc.so", RTLD_LAZY);
+ if (!libc) {
+ LOG("Failed to open libc.so");
+ return -1;
+ }
+ system_property_get* func = (system_property_get*)
+ dlsym(libc, "__system_property_get");
+ int ret = -1;
+ if (func) {
+ ret = func(name, value);
+ }
+ dlclose(libc);
+ return ret;
+}
+#endif
+#endif
+
+static int
+get_android_version(void)
+{
+ char version_string[PROP_VALUE_MAX];
+
+ memset(version_string, 0, PROP_VALUE_MAX);
+
+ int len = __system_property_get("ro.build.version.sdk", version_string);
+ if (len <= 0) {
+ LOG("Failed to get Android version!\n");
+ return len;
+ }
+
+ int version = (int)strtol(version_string, NULL, 10);
+ LOG("%d", version);
+ return version;
+}
+#endif
+
+/*static*/ int
+opensl_init(cubeb ** context, char const * context_name)
+{
+ cubeb * ctx;
+
+#if defined(__ANDROID__)
+ int android_version = get_android_version();
+ if (android_version > 0 && android_version <= ANDROID_VERSION_GINGERBREAD_MR1) {
+ // Don't even attempt to run on Gingerbread and lower
+ return CUBEB_ERROR;
+ }
+#endif
+
+ *context = NULL;
+
+ ctx = calloc(1, sizeof(*ctx));
+ assert(ctx);
+
+ ctx->ops = &opensl_ops;
+
+ ctx->lib = dlopen("libOpenSLES.so", RTLD_LAZY);
+ ctx->libmedia = dlopen("libmedia.so", RTLD_LAZY);
+ if (!ctx->lib || !ctx->libmedia) {
+ free(ctx);
+ return CUBEB_ERROR;
+ }
+
+ /* Get the latency, in ms, from AudioFlinger */
+ /* status_t AudioSystem::getOutputLatency(uint32_t* latency,
+ * audio_stream_type_t streamType) */
+ /* First, try the most recent signature. */
+ ctx->get_output_latency =
+ dlsym(ctx->libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPj19audio_stream_type_t");
+ if (!ctx->get_output_latency) {
+ /* in case of failure, try the legacy version. */
+ /* status_t AudioSystem::getOutputLatency(uint32_t* latency,
+ * int streamType) */
+ ctx->get_output_latency =
+ dlsym(ctx->libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPji");
+ if (!ctx->get_output_latency) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+ }
+
+ typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
+ SLuint32,
+ const SLEngineOption *,
+ SLuint32,
+ const SLInterfaceID *,
+ const SLboolean *);
+ slCreateEngine_t f_slCreateEngine =
+ (slCreateEngine_t)dlsym(ctx->lib, "slCreateEngine");
+ SLInterfaceID SL_IID_ENGINE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ENGINE");
+ SLInterfaceID SL_IID_OUTPUTMIX = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_OUTPUTMIX");
+ ctx->SL_IID_VOLUME = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_VOLUME");
+ ctx->SL_IID_BUFFERQUEUE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_BUFFERQUEUE");
+#if defined(__ANDROID__)
+ ctx->SL_IID_ANDROIDCONFIGURATION = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ANDROIDCONFIGURATION");
+#endif
+ ctx->SL_IID_PLAY = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_PLAY");
+ if (!f_slCreateEngine ||
+ !SL_IID_ENGINE ||
+ !SL_IID_OUTPUTMIX ||
+ !ctx->SL_IID_BUFFERQUEUE ||
+#if defined(__ANDROID__)
+ !ctx->SL_IID_ANDROIDCONFIGURATION ||
+#endif
+ !ctx->SL_IID_PLAY) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ const SLEngineOption opt[] = {{SL_ENGINEOPTION_THREADSAFE, SL_BOOLEAN_TRUE}};
+
+ SLresult res;
+ res = cubeb_get_sles_engine(&ctx->engObj, 1, opt, 0, NULL, NULL);
+
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ res = cubeb_realize_sles_engine(ctx->engObj);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ res = (*ctx->engObj)->GetInterface(ctx->engObj, SL_IID_ENGINE, &ctx->eng);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ const SLInterfaceID idsom[] = {SL_IID_OUTPUTMIX};
+ const SLboolean reqom[] = {SL_BOOLEAN_TRUE};
+ res = (*ctx->eng)->CreateOutputMix(ctx->eng, &ctx->outmixObj, 1, idsom, reqom);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ res = (*ctx->outmixObj)->Realize(ctx->outmixObj, SL_BOOLEAN_FALSE);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_destroy(ctx);
+ return CUBEB_ERROR;
+ }
+
+ *context = ctx;
+
+ return CUBEB_OK;
+}
+
+static char const *
+opensl_get_backend_id(cubeb * ctx)
+{
+ return "opensl";
+}
+
+static int
+opensl_get_max_channel_count(cubeb * ctx, uint32_t * max_channels)
+{
+ assert(ctx && max_channels);
+ /* The android mixer handles up to two channels, see
+ http://androidxref.com/4.2.2_r1/xref/frameworks/av/services/audioflinger/AudioFlinger.h#67 */
+ *max_channels = 2;
+
+ return CUBEB_OK;
+}
+
+static int
+opensl_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate)
+{
+ /* https://android.googlesource.com/platform/ndk.git/+/master/docs/opensles/index.html
+ * We don't want to deal with JNI here (and we don't have Java on b2g anyways),
+ * so we just dlopen the library and get the two symbols we need. */
+ int r;
+ void * libmedia;
+ uint32_t (*get_primary_output_samplingrate)();
+ uint32_t (*get_output_samplingrate)(int * samplingRate, int streamType);
+
+ libmedia = dlopen("libmedia.so", RTLD_LAZY);
+ if (!libmedia) {
+ return CUBEB_ERROR;
+ }
+
+ /* uint32_t AudioSystem::getPrimaryOutputSamplingRate(void) */
+ get_primary_output_samplingrate =
+ dlsym(libmedia, "_ZN7android11AudioSystem28getPrimaryOutputSamplingRateEv");
+ if (!get_primary_output_samplingrate) {
+ /* fallback to
+ * status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
+ * if we cannot find getPrimaryOutputSamplingRate. */
+ get_output_samplingrate =
+ dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPj19audio_stream_type_t");
+ if (!get_output_samplingrate) {
+ /* Another signature exists, with a int instead of an audio_stream_type_t */
+ get_output_samplingrate =
+ dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPii");
+ if (!get_output_samplingrate) {
+ dlclose(libmedia);
+ return CUBEB_ERROR;
+ }
+ }
+ }
+
+ if (get_primary_output_samplingrate) {
+ *rate = get_primary_output_samplingrate();
+ } else {
+ /* We don't really know about the type, here, so we just pass music. */
+ r = get_output_samplingrate((int *) rate, AUDIO_STREAM_TYPE_MUSIC);
+ if (r) {
+ dlclose(libmedia);
+ return CUBEB_ERROR;
+ }
+ }
+
+ dlclose(libmedia);
+
+ /* Depending on which method we called above, we can get a zero back, yet have
+ * a non-error return value, especially if the audio system is not
+ * ready/shutting down (i.e. when we can't get our hand on the AudioFlinger
+ * thread). */
+ if (*rate == 0) {
+ return CUBEB_ERROR;
+ }
+
+ return CUBEB_OK;
+}
+
+static int
+opensl_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames)
+{
+ /* https://android.googlesource.com/platform/ndk.git/+/master/docs/opensles/index.html
+ * We don't want to deal with JNI here (and we don't have Java on b2g anyways),
+ * so we just dlopen the library and get the two symbols we need. */
+
+ int r;
+ void * libmedia;
+ size_t (*get_primary_output_frame_count)(void);
+ int (*get_output_frame_count)(size_t * frameCount, int streamType);
+ uint32_t primary_sampling_rate;
+ size_t primary_buffer_size;
+
+ r = opensl_get_preferred_sample_rate(ctx, &primary_sampling_rate);
+
+ if (r) {
+ return CUBEB_ERROR;
+ }
+
+ libmedia = dlopen("libmedia.so", RTLD_LAZY);
+ if (!libmedia) {
+ return CUBEB_ERROR;
+ }
+
+ /* JB variant */
+ /* size_t AudioSystem::getPrimaryOutputFrameCount(void) */
+ get_primary_output_frame_count =
+ dlsym(libmedia, "_ZN7android11AudioSystem26getPrimaryOutputFrameCountEv");
+ if (!get_primary_output_frame_count) {
+ /* ICS variant */
+ /* status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) */
+ get_output_frame_count =
+ dlsym(libmedia, "_ZN7android11AudioSystem19getOutputFrameCountEPii");
+ if (!get_output_frame_count) {
+ dlclose(libmedia);
+ return CUBEB_ERROR;
+ }
+ }
+
+ if (get_primary_output_frame_count) {
+ primary_buffer_size = get_primary_output_frame_count();
+ } else {
+ if (get_output_frame_count(&primary_buffer_size, params.stream_type) != 0) {
+ return CUBEB_ERROR;
+ }
+ }
+
+ /* To get a fast track in Android's mixer, we need to be at the native
+ * samplerate, which is device dependant. Some devices might be able to
+ * resample when playing a fast track, but it's pretty rare. */
+ *latency_frames = NBUFS * primary_buffer_size;
+
+ dlclose(libmedia);
+
+ return CUBEB_OK;
+}
+
+static void
+opensl_destroy(cubeb * ctx)
+{
+ if (ctx->outmixObj)
+ (*ctx->outmixObj)->Destroy(ctx->outmixObj);
+ if (ctx->engObj)
+ cubeb_destroy_sles_engine(&ctx->engObj);
+ dlclose(ctx->lib);
+ dlclose(ctx->libmedia);
+ free(ctx);
+}
+
+static void opensl_stream_destroy(cubeb_stream * stm);
+
+static int
+opensl_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name,
+ cubeb_devid input_device,
+ cubeb_stream_params * input_stream_params,
+ cubeb_devid output_device,
+ cubeb_stream_params * output_stream_params,
+ unsigned int latency_frames,
+ cubeb_data_callback data_callback, cubeb_state_callback state_callback,
+ void * user_ptr)
+{
+ cubeb_stream * stm;
+
+ assert(ctx);
+ assert(!input_stream_params && "not supported");
+ if (input_device || output_device) {
+ /* Device selection not yet implemented. */
+ return CUBEB_ERROR_DEVICE_UNAVAILABLE;
+ }
+
+ *stream = NULL;
+
+ SLDataFormat_PCM format;
+
+ format.formatType = SL_DATAFORMAT_PCM;
+ format.numChannels = output_stream_params->channels;
+ // samplesPerSec is in milliHertz
+ format.samplesPerSec = output_stream_params->rate * 1000;
+ format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+ format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+ format.channelMask = output_stream_params->channels == 1 ?
+ SL_SPEAKER_FRONT_CENTER :
+ SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+
+ switch (output_stream_params->format) {
+ case CUBEB_SAMPLE_S16LE:
+ format.endianness = SL_BYTEORDER_LITTLEENDIAN;
+ break;
+ case CUBEB_SAMPLE_S16BE:
+ format.endianness = SL_BYTEORDER_BIGENDIAN;
+ break;
+ default:
+ return CUBEB_ERROR_INVALID_FORMAT;
+ }
+
+ stm = calloc(1, sizeof(*stm));
+ assert(stm);
+
+ stm->context = ctx;
+ stm->data_callback = data_callback;
+ stm->state_callback = state_callback;
+ stm->user_ptr = user_ptr;
+
+ stm->inputrate = output_stream_params->rate;
+ stm->latency = latency_frames;
+ stm->stream_type = output_stream_params->stream_type;
+ stm->framesize = output_stream_params->channels * sizeof(int16_t);
+ stm->lastPosition = -1;
+ stm->lastPositionTimeStamp = 0;
+ stm->lastCompensativePosition = -1;
+
+ int r = pthread_mutex_init(&stm->mutex, NULL);
+ assert(r == 0);
+
+ SLDataLocator_BufferQueue loc_bufq;
+ loc_bufq.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
+ loc_bufq.numBuffers = NBUFS;
+ SLDataSource source;
+ source.pLocator = &loc_bufq;
+ source.pFormat = &format;
+
+ SLDataLocator_OutputMix loc_outmix;
+ loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+ loc_outmix.outputMix = ctx->outmixObj;
+ SLDataSink sink;
+ sink.pLocator = &loc_outmix;
+ sink.pFormat = NULL;
+
+#if defined(__ANDROID__)
+ const SLInterfaceID ids[] = {ctx->SL_IID_BUFFERQUEUE,
+ ctx->SL_IID_VOLUME,
+ ctx->SL_IID_ANDROIDCONFIGURATION};
+ const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+#else
+ const SLInterfaceID ids[] = {ctx->SL_IID_BUFFERQUEUE, ctx->SL_IID_VOLUME};
+ const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+#endif
+ assert(NELEMS(ids) == NELEMS(req));
+
+ uint32_t preferred_sampling_rate = stm->inputrate;
+#if defined(__ANDROID__)
+ if (get_android_version() >= ANDROID_VERSION_MARSHMALLOW) {
+ // Reset preferred samping rate to trigger fallback to native sampling rate.
+ preferred_sampling_rate = 0;
+ if (opensl_get_min_latency(ctx, *output_stream_params, &latency_frames) != CUBEB_OK) {
+ // Default to AudioFlinger's advertised fast track latency of 10ms.
+ latency_frames = 440;
+ }
+ stm->latency = latency_frames;
+ }
+#endif
+
+ SLresult res = SL_RESULT_CONTENT_UNSUPPORTED;
+ if (preferred_sampling_rate) {
+ res = (*ctx->eng)->CreateAudioPlayer(ctx->eng, &stm->playerObj, &source,
+ &sink, NELEMS(ids), ids, req);
+ }
+
+ // Sample rate not supported? Try again with primary sample rate!
+ if (res == SL_RESULT_CONTENT_UNSUPPORTED) {
+ if (opensl_get_preferred_sample_rate(ctx, &preferred_sampling_rate)) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ format.samplesPerSec = preferred_sampling_rate * 1000;
+ res = (*ctx->eng)->CreateAudioPlayer(ctx->eng, &stm->playerObj,
+ &source, &sink, NELEMS(ids), ids, req);
+ }
+
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ stm->outputrate = preferred_sampling_rate;
+ stm->bytespersec = stm->outputrate * stm->framesize;
+ stm->queuebuf_len = stm->framesize * latency_frames / NBUFS;
+ // round up to the next multiple of stm->framesize, if needed.
+ if (stm->queuebuf_len % stm->framesize) {
+ stm->queuebuf_len += stm->framesize - (stm->queuebuf_len % stm->framesize);
+ }
+
+ cubeb_stream_params params = *output_stream_params;
+ params.rate = preferred_sampling_rate;
+
+ stm->resampler = cubeb_resampler_create(stm, NULL, &params,
+ output_stream_params->rate,
+ data_callback,
+ user_ptr,
+ CUBEB_RESAMPLER_QUALITY_DEFAULT);
+
+ if (!stm->resampler) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ int i;
+ for (i = 0; i < NBUFS; i++) {
+ stm->queuebuf[i] = malloc(stm->queuebuf_len);
+ assert(stm->queuebuf[i]);
+ }
+
+#if defined(__ANDROID__)
+ SLuint32 stream_type = convert_stream_type_to_sl_stream(output_stream_params->stream_type);
+ if (stream_type != 0xFFFFFFFF) {
+ SLAndroidConfigurationItf playerConfig;
+ res = (*stm->playerObj)->GetInterface(stm->playerObj,
+ ctx->SL_IID_ANDROIDCONFIGURATION, &playerConfig);
+ res = (*playerConfig)->SetConfiguration(playerConfig,
+ SL_ANDROID_KEY_STREAM_TYPE, &stream_type, sizeof(SLint32));
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+ }
+#endif
+
+ res = (*stm->playerObj)->Realize(stm->playerObj, SL_BOOLEAN_FALSE);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_PLAY, &stm->play);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_BUFFERQUEUE,
+ &stm->bufq);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_VOLUME,
+ &stm->volume);
+
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ res = (*stm->play)->RegisterCallback(stm->play, play_callback, stm);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ // Work around wilhelm/AudioTrack badness, bug 1221228
+ (*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)0);
+
+ res = (*stm->play)->SetCallbackEventsMask(stm->play, (SLuint32)SL_PLAYEVENT_HEADATMARKER);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ res = (*stm->bufq)->RegisterCallback(stm->bufq, bufferqueue_callback, stm);
+ if (res != SL_RESULT_SUCCESS) {
+ opensl_stream_destroy(stm);
+ return CUBEB_ERROR;
+ }
+
+ {
+ // Enqueue a silent frame so once the player becomes playing, the frame
+ // will be consumed and kick off the buffer queue callback.
+ // Note the duration of a single frame is less than 1ms. We don't bother
+ // adjusting the playback position.
+ uint8_t *buf = stm->queuebuf[stm->queuebuf_idx++];
+ memset(buf, 0, stm->framesize);
+ res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->framesize);
+ assert(res == SL_RESULT_SUCCESS);
+ }
+
+ *stream = stm;
+ return CUBEB_OK;
+}
+
+static void
+opensl_stream_destroy(cubeb_stream * stm)
+{
+ if (stm->playerObj)
+ (*stm->playerObj)->Destroy(stm->playerObj);
+ int i;
+ for (i = 0; i < NBUFS; i++) {
+ free(stm->queuebuf[i]);
+ }
+ pthread_mutex_destroy(&stm->mutex);
+
+ cubeb_resampler_destroy(stm->resampler);
+
+ free(stm);
+}
+
+static int
+opensl_stream_start(cubeb_stream * stm)
+{
+ SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PLAYING);
+ if (res != SL_RESULT_SUCCESS)
+ return CUBEB_ERROR;
+ stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED);
+ return CUBEB_OK;
+}
+
+static int
+opensl_stream_stop(cubeb_stream * stm)
+{
+ SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
+ if (res != SL_RESULT_SUCCESS)
+ return CUBEB_ERROR;
+ stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED);
+ return CUBEB_OK;
+}
+
+static int
+opensl_stream_get_position(cubeb_stream * stm, uint64_t * position)
+{
+ SLmillisecond msec;
+ uint64_t samplerate;
+ SLresult res;
+ int r;
+ uint32_t mixer_latency;
+ uint32_t compensation_msec = 0;
+
+ res = (*stm->play)->GetPosition(stm->play, &msec);
+ if (res != SL_RESULT_SUCCESS)
+ return CUBEB_ERROR;
+
+ struct timespec t;
+ clock_gettime(CLOCK_MONOTONIC, &t);
+ if(stm->lastPosition == msec) {
+ compensation_msec =
+ (t.tv_sec*1000000000LL + t.tv_nsec - stm->lastPositionTimeStamp) / 1000000;
+ } else {
+ stm->lastPositionTimeStamp = t.tv_sec*1000000000LL + t.tv_nsec;
+ stm->lastPosition = msec;
+ }
+
+ samplerate = stm->inputrate;
+
+ r = stm->context->get_output_latency(&mixer_latency, stm->stream_type);
+ if (r) {
+ return CUBEB_ERROR;
+ }
+
+ pthread_mutex_lock(&stm->mutex);
+ int64_t maximum_position = stm->written * (int64_t)stm->inputrate / stm->outputrate;
+ pthread_mutex_unlock(&stm->mutex);
+ assert(maximum_position >= 0);
+
+ if (msec > mixer_latency) {
+ int64_t unadjusted_position;
+ if (stm->lastCompensativePosition > msec + compensation_msec) {
+ // Over compensation, use lastCompensativePosition.
+ unadjusted_position =
+ samplerate * (stm->lastCompensativePosition - mixer_latency) / 1000;
+ } else {
+ unadjusted_position =
+ samplerate * (msec - mixer_latency + compensation_msec) / 1000;
+ stm->lastCompensativePosition = msec + compensation_msec;
+ }
+ *position = unadjusted_position < maximum_position ?
+ unadjusted_position : maximum_position;
+ } else {
+ *position = 0;
+ }
+ return CUBEB_OK;
+}
+
+int
+opensl_stream_get_latency(cubeb_stream * stm, uint32_t * latency)
+{
+ int r;
+ uint32_t mixer_latency; // The latency returned by AudioFlinger is in ms.
+
+ /* audio_stream_type_t is an int, so this is okay. */
+ r = stm->context->get_output_latency(&mixer_latency, stm->stream_type);
+ if (r) {
+ return CUBEB_ERROR;
+ }
+
+ *latency = stm->latency * stm->inputrate / 1000 + // OpenSL latency
+ mixer_latency * stm->inputrate / 1000; // AudioFlinger latency
+
+ return CUBEB_OK;
+}
+
+int
+opensl_stream_set_volume(cubeb_stream * stm, float volume)
+{
+ SLresult res;
+ SLmillibel max_level, millibels;
+ float unclamped_millibels;
+
+ res = (*stm->volume)->GetMaxVolumeLevel(stm->volume, &max_level);
+
+ if (res != SL_RESULT_SUCCESS) {
+ return CUBEB_ERROR;
+ }
+
+ /* millibels are 100*dB, so the conversion from the volume's linear amplitude
+ * is 100 * 20 * log(volume). However we clamp the resulting value before
+ * passing it to lroundf() in order to prevent it from silently returning an
+ * erroneous value when the unclamped value exceeds the size of a long. */
+ unclamped_millibels = 100.0f * 20.0f * log10f(fmaxf(volume, 0.0f));
+ unclamped_millibels = fmaxf(unclamped_millibels, SL_MILLIBEL_MIN);
+ unclamped_millibels = fminf(unclamped_millibels, max_level);
+
+ millibels = lroundf(unclamped_millibels);
+
+ res = (*stm->volume)->SetVolumeLevel(stm->volume, millibels);
+
+ if (res != SL_RESULT_SUCCESS) {
+ return CUBEB_ERROR;
+ }
+ return CUBEB_OK;
+}
+
+static struct cubeb_ops const opensl_ops = {
+ .init = opensl_init,
+ .get_backend_id = opensl_get_backend_id,
+ .get_max_channel_count = opensl_get_max_channel_count,
+ .get_min_latency = opensl_get_min_latency,
+ .get_preferred_sample_rate = opensl_get_preferred_sample_rate,
+ .enumerate_devices = NULL,
+ .destroy = opensl_destroy,
+ .stream_init = opensl_stream_init,
+ .stream_destroy = opensl_stream_destroy,
+ .stream_start = opensl_stream_start,
+ .stream_stop = opensl_stream_stop,
+ .stream_get_position = opensl_stream_get_position,
+ .stream_get_latency = opensl_stream_get_latency,
+ .stream_set_volume = opensl_stream_set_volume,
+ .stream_set_panning = NULL,
+ .stream_get_current_device = NULL,
+ .stream_device_destroy = NULL,
+ .stream_register_device_changed_callback = NULL,
+ .register_device_collection_changed = NULL
+};