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265 lines
8.7 KiB
C
265 lines
8.7 KiB
C
/* GStreamer
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* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-openslessink
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* @see_also: openslessrc
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*
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* This element renders raw audio samples using the OpenSL ES API in Android OS.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v filesrc location=music.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! opeslessink
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* ]| Play an Ogg/Vorbis file.
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* </refsect2>
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include "opensles.h"
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#include "openslessink.h"
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GST_DEBUG_CATEGORY_STATIC (opensles_sink_debug);
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#define GST_CAT_DEFAULT opensles_sink_debug
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enum
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{
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PROP_0,
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PROP_VOLUME,
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PROP_MUTE,
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PROP_LAST
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};
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#define DEFAULT_VOLUME 1.0
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#define DEFAULT_MUTE FALSE
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/* According to Android's NDK doc the following are the supported rates */
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#define RATES "8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100"
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/* 48000 Hz is also claimed to be supported but the AudioFlinger downsampling
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* doesn't seems to work properly so we relay GStreamer audioresample element
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* to cope with this samplerate. */
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (U8) "}, "
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"rate = (int) { " RATES "}, " "channels = (int) [1, 2]")
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);
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#define _do_init \
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GST_DEBUG_CATEGORY_INIT (opensles_sink_debug, "opensles_sink", 0, \
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"OpenSL ES Sink");
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#define parent_class gst_opensles_sink_parent_class
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G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSink, gst_opensles_sink,
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GST_TYPE_AUDIO_BASE_SINK, _do_init);
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static GstAudioRingBuffer *
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gst_opensles_sink_create_ringbuffer (GstAudioBaseSink * base)
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{
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GstOpenSLESSink *sink = GST_OPENSLES_SINK (base);
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GstAudioRingBuffer *rb;
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rb = gst_opensles_ringbuffer_new (RB_MODE_SINK_PCM);
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gst_opensles_ringbuffer_set_volume (rb, sink->volume);
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gst_opensles_ringbuffer_set_mute (rb, sink->mute);
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return rb;
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}
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#define AUDIO_OUTPUT_DESC_FORMAT \
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"deviceName: %s deviceConnection: %d deviceScope: %d deviceLocation: %d " \
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"isForTelephony: %d minSampleRate: %d maxSampleRate: %d " \
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"isFreqRangeContinuous: %d maxChannels: %d"
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#define AUDIO_OUTPUT_DESC_ARGS(aod) \
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(gchar*) (aod)->pDeviceName, (gint) (aod)->deviceConnection, \
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(gint) (aod)->deviceScope, (gint) (aod)->deviceLocation, \
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(gint) (aod)->isForTelephony, (gint) (aod)->minSampleRate, \
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(gint) (aod)->maxSampleRate, (gint) (aod)->isFreqRangeContinuous, \
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(gint) (aod)->maxChannels
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static gboolean
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_opensles_query_capabilities (GstOpenSLESSink * sink)
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{
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gboolean res = FALSE;
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SLresult result;
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SLObjectItf engineObject = NULL;
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SLAudioIODeviceCapabilitiesItf audioIODeviceCapabilities;
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SLint32 i, j, numOutputs = MAX_NUMBER_OUTPUT_DEVICES;
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SLuint32 outputDeviceIDs[MAX_NUMBER_OUTPUT_DEVICES];
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SLAudioOutputDescriptor audioOutputDescriptor;
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/* Create and realize engine */
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engineObject = gst_opensles_get_engine ();
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if (!engineObject) {
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GST_ERROR_OBJECT (sink, "Getting engine failed");
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goto beach;
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}
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/* Get the engine interface, which is needed in order to create other objects */
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result = (*engineObject)->GetInterface (engineObject,
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SL_IID_AUDIOIODEVICECAPABILITIES, &audioIODeviceCapabilities);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (sink,
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"engine.GetInterface(IODeviceCapabilities) failed(0x%08x)",
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(guint32) result);
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goto beach;
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}
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/* Query the list of available audio outputs */
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result = (*audioIODeviceCapabilities)->GetAvailableAudioOutputs
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(audioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (sink,
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"IODeviceCapabilities.GetAvailableAudioOutputs failed(0x%08x)",
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(guint32) result);
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goto beach;
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}
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GST_DEBUG_OBJECT (sink, "Found %d output devices", (gint32) numOutputs);
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for (i = 0; i < numOutputs; i++) {
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result = (*audioIODeviceCapabilities)->QueryAudioOutputCapabilities
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(audioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor);
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if (result != SL_RESULT_SUCCESS) {
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GST_ERROR_OBJECT (sink,
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"IODeviceCapabilities.QueryAudioOutputCapabilities failed(0x%08x)",
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(guint32) result);
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continue;
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}
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GST_DEBUG_OBJECT (sink, " ID: %08x " AUDIO_OUTPUT_DESC_FORMAT,
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(guint) outputDeviceIDs[i],
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AUDIO_OUTPUT_DESC_ARGS (&audioOutputDescriptor));
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GST_DEBUG_OBJECT (sink, " Found %d supported sample rated",
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audioOutputDescriptor.numOfSamplingRatesSupported);
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for (j = 0; j < audioOutputDescriptor.numOfSamplingRatesSupported; j++) {
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GST_DEBUG_OBJECT (sink, " %d Hz",
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(gint) audioOutputDescriptor.samplingRatesSupported[j]);
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}
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}
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res = TRUE;
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beach:
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/* Destroy the engine object */
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if (engineObject) {
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gst_opensles_release_engine (engineObject);
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}
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return res;
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}
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static void
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gst_opensles_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
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GstAudioRingBuffer *rb = GST_AUDIO_BASE_SINK (sink)->ringbuffer;
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switch (prop_id) {
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case PROP_VOLUME:
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sink->volume = g_value_get_double (value);
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if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
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gst_opensles_ringbuffer_set_volume (rb, sink->volume);
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}
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break;
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case PROP_MUTE:
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sink->mute = g_value_get_boolean (value);
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if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
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gst_opensles_ringbuffer_set_mute (rb, sink->mute);
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}
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_opensles_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
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switch (prop_id) {
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case PROP_VOLUME:
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g_value_set_double (value, sink->volume);
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break;
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case PROP_MUTE:
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g_value_set_boolean (value, sink->mute);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_opensles_sink_class_init (GstOpenSLESSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioBaseSinkClass *gstbaseaudiosink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
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gobject_class->set_property = gst_opensles_sink_set_property;
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gobject_class->get_property = gst_opensles_sink_get_property;
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g_object_class_install_property (gobject_class, PROP_VOLUME,
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g_param_spec_double ("volume", "Volume", "Volume of this stream",
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0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MUTE,
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g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
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DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Sink",
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"Sink/Audio",
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"Output sound using the OpenSL ES APIs",
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"Josep Torra <support@fluendo.com>");
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gstbaseaudiosink_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_opensles_sink_create_ringbuffer);
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}
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static void
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gst_opensles_sink_init (GstOpenSLESSink * sink)
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{
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sink->volume = DEFAULT_VOLUME;
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sink->mute = DEFAULT_MUTE;
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_opensles_query_capabilities (sink);
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gst_audio_base_sink_set_provide_clock (GST_AUDIO_BASE_SINK (sink), TRUE);
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/* Override some default values to fit on the AudioFlinger behaviour of
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* processing 20ms buffers as minimum buffer size. */
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GST_AUDIO_BASE_SINK (sink)->buffer_time = 400000;
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GST_AUDIO_BASE_SINK (sink)->latency_time = 20000;
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}
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