blob: 236f8b3c6fc0667a0469c6130453b591f7bb5278 [file] [log] [blame]
/* GStreamer
* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-openslessink
* @see_also: openslessrc
*
* This element renders raw audio samples using the OpenSL ES API in Android OS.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=music.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! opeslessink
* ]| Play an Ogg/Vorbis file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "opensles.h"
#include "openslessink.h"
GST_DEBUG_CATEGORY_STATIC (opensles_sink_debug);
#define GST_CAT_DEFAULT opensles_sink_debug
enum
{
PROP_0,
PROP_VOLUME,
PROP_MUTE,
PROP_LAST
};
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
/* According to Android's NDK doc the following are the supported rates */
#define RATES "8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100"
/* 48000 Hz is also claimed to be supported but the AudioFlinger downsampling
* doesn't seems to work properly so we relay GStreamer audioresample element
* to cope with this samplerate. */
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (U8) "}, "
"rate = (int) { " RATES "}, " "channels = (int) [1, 2], "
"layout = (string) interleaved")
);
#define _do_init \
GST_DEBUG_CATEGORY_INIT (opensles_sink_debug, "openslessink", 0, \
"OpenSLES Sink");
#define parent_class gst_opensles_sink_parent_class
G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSink, gst_opensles_sink,
GST_TYPE_AUDIO_BASE_SINK, _do_init);
static GstAudioRingBuffer *
gst_opensles_sink_create_ringbuffer (GstAudioBaseSink * base)
{
GstOpenSLESSink *sink = GST_OPENSLES_SINK (base);
GstAudioRingBuffer *rb;
rb = gst_opensles_ringbuffer_new (RB_MODE_SINK_PCM);
gst_opensles_ringbuffer_set_volume (rb, sink->volume);
gst_opensles_ringbuffer_set_mute (rb, sink->mute);
return rb;
}
#define AUDIO_OUTPUT_DESC_FORMAT \
"deviceName: %s deviceConnection: %d deviceScope: %d deviceLocation: %d " \
"isForTelephony: %d minSampleRate: %d maxSampleRate: %d " \
"isFreqRangeContinuous: %d maxChannels: %d"
#define AUDIO_OUTPUT_DESC_ARGS(aod) \
(gchar*) (aod)->pDeviceName, (gint) (aod)->deviceConnection, \
(gint) (aod)->deviceScope, (gint) (aod)->deviceLocation, \
(gint) (aod)->isForTelephony, (gint) (aod)->minSampleRate, \
(gint) (aod)->maxSampleRate, (gint) (aod)->isFreqRangeContinuous, \
(gint) (aod)->maxChannels
static gboolean
_opensles_query_capabilities (GstOpenSLESSink * sink)
{
gboolean res = FALSE;
SLresult result;
SLObjectItf engineObject = NULL;
SLAudioIODeviceCapabilitiesItf audioIODeviceCapabilities;
SLint32 i, j, numOutputs = MAX_NUMBER_OUTPUT_DEVICES;
SLuint32 outputDeviceIDs[MAX_NUMBER_OUTPUT_DEVICES];
SLAudioOutputDescriptor audioOutputDescriptor;
/* Create and realize engine */
engineObject = gst_opensles_get_engine ();
if (!engineObject) {
GST_ERROR_OBJECT (sink, "Getting engine failed");
goto beach;
}
/* Get the engine interface, which is needed in order to create other objects */
result = (*engineObject)->GetInterface (engineObject,
SL_IID_AUDIOIODEVICECAPABILITIES, &audioIODeviceCapabilities);
if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
GST_LOG_OBJECT (sink,
"engine.GetInterface(IODeviceCapabilities) unsupported(0x%08x)",
(guint32) result);
goto beach;
} else if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (sink,
"engine.GetInterface(IODeviceCapabilities) failed(0x%08x)",
(guint32) result);
goto beach;
}
/* Query the list of available audio outputs */
result = (*audioIODeviceCapabilities)->GetAvailableAudioOutputs
(audioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
GST_LOG_OBJECT (sink,
"IODeviceCapabilities.GetAvailableAudioOutputs unsupported(0x%08x)",
(guint32) result);
goto beach;
} else if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (sink,
"IODeviceCapabilities.GetAvailableAudioOutputs failed(0x%08x)",
(guint32) result);
goto beach;
}
GST_DEBUG_OBJECT (sink, "Found %d output devices", (gint32) numOutputs);
for (i = 0; i < numOutputs; i++) {
result = (*audioIODeviceCapabilities)->QueryAudioOutputCapabilities
(audioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor);
if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
GST_LOG_OBJECT (sink,
"IODeviceCapabilities.QueryAudioOutputCapabilities unsupported(0x%08x)",
(guint32) result);
continue;
} else if (result != SL_RESULT_SUCCESS) {
GST_ERROR_OBJECT (sink,
"IODeviceCapabilities.QueryAudioOutputCapabilities failed(0x%08x)",
(guint32) result);
continue;
}
GST_DEBUG_OBJECT (sink, " ID: %08x " AUDIO_OUTPUT_DESC_FORMAT,
(guint) outputDeviceIDs[i],
AUDIO_OUTPUT_DESC_ARGS (&audioOutputDescriptor));
GST_DEBUG_OBJECT (sink, " Found %d supported sample rated",
audioOutputDescriptor.numOfSamplingRatesSupported);
for (j = 0; j < audioOutputDescriptor.numOfSamplingRatesSupported; j++) {
GST_DEBUG_OBJECT (sink, " %d Hz",
(gint) audioOutputDescriptor.samplingRatesSupported[j]);
}
}
res = TRUE;
beach:
/* Destroy the engine object */
if (engineObject) {
gst_opensles_release_engine (engineObject);
}
return res;
}
static void
gst_opensles_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
GstAudioRingBuffer *rb = GST_AUDIO_BASE_SINK (sink)->ringbuffer;
switch (prop_id) {
case PROP_VOLUME:
sink->volume = g_value_get_double (value);
if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
gst_opensles_ringbuffer_set_volume (rb, sink->volume);
}
break;
case PROP_MUTE:
sink->mute = g_value_get_boolean (value);
if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
gst_opensles_ringbuffer_set_mute (rb, sink->mute);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_opensles_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
g_value_set_double (value, sink->volume);
break;
case PROP_MUTE:
g_value_set_boolean (value, sink->mute);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_opensles_sink_class_init (GstOpenSLESSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioBaseSinkClass *gstbaseaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
gobject_class->set_property = gst_opensles_sink_set_property;
gobject_class->get_property = gst_opensles_sink_get_property;
g_object_class_install_property (gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume of this stream",
0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Sink",
"Sink/Audio",
"Output sound using the OpenSL ES APIs",
"Josep Torra <support@fluendo.com>");
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_opensles_sink_create_ringbuffer);
}
static void
gst_opensles_sink_init (GstOpenSLESSink * sink)
{
sink->volume = DEFAULT_VOLUME;
sink->mute = DEFAULT_MUTE;
_opensles_query_capabilities (sink);
gst_audio_base_sink_set_provide_clock (GST_AUDIO_BASE_SINK (sink), TRUE);
/* Override some default values to fit on the AudioFlinger behaviour of
* processing 20ms buffers as minimum buffer size. */
GST_AUDIO_BASE_SINK (sink)->buffer_time = 200000;
GST_AUDIO_BASE_SINK (sink)->latency_time = 20000;
}