1 /* GStreamer
2 * Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
3 *
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
8 *
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
13 *
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 /**
21 * SECTION:element-openslessink
22 * @title: openslessink
23 * @see_also: openslessrc
24 *
25 * This element renders raw audio samples using the OpenSL ES API in Android OS.
26 *
27 * ## Example pipelines
28 * |[
29 * gst-launch-1.0 -v filesrc location=music.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! opeslessink
30 * ]| Play an Ogg/Vorbis file.
31 *
32 */
33
34 #ifdef HAVE_CONFIG_H
35 # include <config.h>
36 #endif
37
38 #include "opensles.h"
39 #include "openslessink.h"
40
41 GST_DEBUG_CATEGORY_STATIC (opensles_sink_debug);
42 #define GST_CAT_DEFAULT opensles_sink_debug
43
44 enum
45 {
46 PROP_0,
47 PROP_VOLUME,
48 PROP_MUTE,
49 PROP_STREAM_TYPE,
50 PROP_LAST
51 };
52
53 #define DEFAULT_VOLUME 1.0
54 #define DEFAULT_MUTE FALSE
55
56 #define DEFAULT_STREAM_TYPE GST_OPENSLES_STREAM_TYPE_NONE
57
58
59 /* According to Android's NDK doc the following are the supported rates */
60 #define RATES "8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100"
61 /* 48000 Hz is also claimed to be supported but the AudioFlinger downsampling
62 * doesn't seems to work properly so we relay GStreamer audioresample element
63 * to cope with this samplerate. */
64
65 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
66 GST_PAD_SINK,
67 GST_PAD_ALWAYS,
68 GST_STATIC_CAPS ("audio/x-raw, "
69 "format = (string) { " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (U8) "}, "
70 "rate = (int) { " RATES "}, " "channels = (int) [1, 2], "
71 "layout = (string) interleaved")
72 );
73
74 #define _do_init \
75 GST_DEBUG_CATEGORY_INIT (opensles_sink_debug, "openslessink", 0, \
76 "OpenSLES Sink");
77 #define parent_class gst_opensles_sink_parent_class
78 G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSink, gst_opensles_sink,
79 GST_TYPE_AUDIO_BASE_SINK, _do_init);
80
81 static GstAudioRingBuffer *
gst_opensles_sink_create_ringbuffer(GstAudioBaseSink * base)82 gst_opensles_sink_create_ringbuffer (GstAudioBaseSink * base)
83 {
84 GstOpenSLESSink *sink = GST_OPENSLES_SINK (base);
85 GstAudioRingBuffer *rb;
86
87 rb = gst_opensles_ringbuffer_new (RB_MODE_SINK_PCM);
88 gst_opensles_ringbuffer_set_volume (rb, sink->volume);
89 gst_opensles_ringbuffer_set_mute (rb, sink->mute);
90
91 GST_OPENSLES_RING_BUFFER (rb)->stream_type = sink->stream_type;
92
93 return rb;
94 }
95
96 #define AUDIO_OUTPUT_DESC_FORMAT \
97 "deviceName: %s deviceConnection: %d deviceScope: %d deviceLocation: %d " \
98 "isForTelephony: %d minSampleRate: %d maxSampleRate: %d " \
99 "isFreqRangeContinuous: %d maxChannels: %d"
100
101 #define AUDIO_OUTPUT_DESC_ARGS(aod) \
102 (gchar*) (aod)->pDeviceName, (gint) (aod)->deviceConnection, \
103 (gint) (aod)->deviceScope, (gint) (aod)->deviceLocation, \
104 (gint) (aod)->isForTelephony, (gint) (aod)->minSampleRate, \
105 (gint) (aod)->maxSampleRate, (gint) (aod)->isFreqRangeContinuous, \
106 (gint) (aod)->maxChannels
107
108 static gboolean
_opensles_query_capabilities(GstOpenSLESSink * sink)109 _opensles_query_capabilities (GstOpenSLESSink * sink)
110 {
111 gboolean res = FALSE;
112 SLresult result;
113 SLObjectItf engineObject = NULL;
114 SLAudioIODeviceCapabilitiesItf audioIODeviceCapabilities;
115 SLint32 i, j, numOutputs = MAX_NUMBER_OUTPUT_DEVICES;
116 SLuint32 outputDeviceIDs[MAX_NUMBER_OUTPUT_DEVICES];
117 SLAudioOutputDescriptor audioOutputDescriptor;
118
119 /* Create and realize engine */
120 engineObject = gst_opensles_get_engine ();
121 if (!engineObject) {
122 GST_ERROR_OBJECT (sink, "Getting engine failed");
123 goto beach;
124 }
125
126 /* Get the engine interface, which is needed in order to create other objects */
127 result = (*engineObject)->GetInterface (engineObject,
128 SL_IID_AUDIOIODEVICECAPABILITIES, &audioIODeviceCapabilities);
129 if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
130 GST_LOG_OBJECT (sink,
131 "engine.GetInterface(IODeviceCapabilities) unsupported(0x%08x)",
132 (guint32) result);
133 goto beach;
134 } else if (result != SL_RESULT_SUCCESS) {
135 GST_ERROR_OBJECT (sink,
136 "engine.GetInterface(IODeviceCapabilities) failed(0x%08x)",
137 (guint32) result);
138 goto beach;
139 }
140
141 /* Query the list of available audio outputs */
142 result = (*audioIODeviceCapabilities)->GetAvailableAudioOutputs
143 (audioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
144 if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
145 GST_LOG_OBJECT (sink,
146 "IODeviceCapabilities.GetAvailableAudioOutputs unsupported(0x%08x)",
147 (guint32) result);
148 goto beach;
149 } else if (result != SL_RESULT_SUCCESS) {
150 GST_ERROR_OBJECT (sink,
151 "IODeviceCapabilities.GetAvailableAudioOutputs failed(0x%08x)",
152 (guint32) result);
153 goto beach;
154 }
155
156 GST_DEBUG_OBJECT (sink, "Found %d output devices", (gint32) numOutputs);
157
158 for (i = 0; i < numOutputs; i++) {
159 result = (*audioIODeviceCapabilities)->QueryAudioOutputCapabilities
160 (audioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor);
161
162 if (result == SL_RESULT_FEATURE_UNSUPPORTED) {
163 GST_LOG_OBJECT (sink,
164 "IODeviceCapabilities.QueryAudioOutputCapabilities unsupported(0x%08x)",
165 (guint32) result);
166 continue;
167 } else if (result != SL_RESULT_SUCCESS) {
168 GST_ERROR_OBJECT (sink,
169 "IODeviceCapabilities.QueryAudioOutputCapabilities failed(0x%08x)",
170 (guint32) result);
171 continue;
172 }
173
174 GST_DEBUG_OBJECT (sink, " ID: %08x " AUDIO_OUTPUT_DESC_FORMAT,
175 (guint) outputDeviceIDs[i],
176 AUDIO_OUTPUT_DESC_ARGS (&audioOutputDescriptor));
177 GST_DEBUG_OBJECT (sink, " Found %d supported sample rated",
178 audioOutputDescriptor.numOfSamplingRatesSupported);
179
180 for (j = 0; j < audioOutputDescriptor.numOfSamplingRatesSupported; j++) {
181 GST_DEBUG_OBJECT (sink, " %d Hz",
182 (gint) audioOutputDescriptor.samplingRatesSupported[j]);
183 }
184 }
185
186 res = TRUE;
187 beach:
188 /* Destroy the engine object */
189 if (engineObject) {
190 gst_opensles_release_engine (engineObject);
191 }
192
193 return res;
194 }
195
196 static void
gst_opensles_sink_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)197 gst_opensles_sink_set_property (GObject * object, guint prop_id,
198 const GValue * value, GParamSpec * pspec)
199 {
200 GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
201 GstAudioRingBuffer *rb = GST_AUDIO_BASE_SINK (sink)->ringbuffer;
202
203 switch (prop_id) {
204 case PROP_VOLUME:
205 sink->volume = g_value_get_double (value);
206 if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
207 gst_opensles_ringbuffer_set_volume (rb, sink->volume);
208 }
209 break;
210 case PROP_MUTE:
211 sink->mute = g_value_get_boolean (value);
212 if (rb && GST_IS_OPENSLES_RING_BUFFER (rb)) {
213 gst_opensles_ringbuffer_set_mute (rb, sink->mute);
214 }
215 break;
216 case PROP_STREAM_TYPE:
217 sink->stream_type = g_value_get_enum (value);
218 break;
219 default:
220 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
221 break;
222 }
223 }
224
225 static void
gst_opensles_sink_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)226 gst_opensles_sink_get_property (GObject * object, guint prop_id,
227 GValue * value, GParamSpec * pspec)
228 {
229 GstOpenSLESSink *sink = GST_OPENSLES_SINK (object);
230 switch (prop_id) {
231 case PROP_VOLUME:
232 g_value_set_double (value, sink->volume);
233 break;
234 case PROP_MUTE:
235 g_value_set_boolean (value, sink->mute);
236 break;
237 case PROP_STREAM_TYPE:
238 g_value_set_enum (value, sink->stream_type);
239 break;
240 default:
241 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
242 break;
243 }
244 }
245
246 static void
gst_opensles_sink_class_init(GstOpenSLESSinkClass * klass)247 gst_opensles_sink_class_init (GstOpenSLESSinkClass * klass)
248 {
249 GObjectClass *gobject_class;
250 GstElementClass *gstelement_class;
251 GstAudioBaseSinkClass *gstbaseaudiosink_class;
252
253 gobject_class = (GObjectClass *) klass;
254 gstelement_class = (GstElementClass *) klass;
255 gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
256
257 gobject_class->set_property = gst_opensles_sink_set_property;
258 gobject_class->get_property = gst_opensles_sink_get_property;
259
260 g_object_class_install_property (gobject_class, PROP_VOLUME,
261 g_param_spec_double ("volume", "Volume", "Volume of this stream",
262 0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
263
264 g_object_class_install_property (gobject_class, PROP_MUTE,
265 g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
266 DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
267
268 g_object_class_install_property (gobject_class, PROP_STREAM_TYPE,
269 g_param_spec_enum ("stream-type", "Stream type",
270 "Stream type that this stream should be tagged with",
271 GST_TYPE_OPENSLES_STREAM_TYPE, DEFAULT_STREAM_TYPE,
272 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
273
274 gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
275
276 gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Sink",
277 "Sink/Audio",
278 "Output sound using the OpenSL ES APIs",
279 "Josep Torra <support@fluendo.com>");
280
281 gstbaseaudiosink_class->create_ringbuffer =
282 GST_DEBUG_FUNCPTR (gst_opensles_sink_create_ringbuffer);
283 }
284
285 static void
gst_opensles_sink_init(GstOpenSLESSink * sink)286 gst_opensles_sink_init (GstOpenSLESSink * sink)
287 {
288 sink->stream_type = DEFAULT_STREAM_TYPE;
289 sink->volume = DEFAULT_VOLUME;
290 sink->mute = DEFAULT_MUTE;
291
292 _opensles_query_capabilities (sink);
293
294 gst_audio_base_sink_set_provide_clock (GST_AUDIO_BASE_SINK (sink), TRUE);
295 /* Override some default values to fit on the AudioFlinger behaviour of
296 * processing 20ms buffers as minimum buffer size. */
297 GST_AUDIO_BASE_SINK (sink)->buffer_time = 200000;
298 GST_AUDIO_BASE_SINK (sink)->latency_time = 20000;
299 }
300