1 /* GStreamer 2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> 3 * 2005 Wim Taymans <wim@fluendo.com> 4 * 5 * gstaudioringbuffer.h: 6 * 7 * This library is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Library General Public 9 * License as published by the Free Software Foundation; either 10 * version 2 of the License, or (at your option) any later version. 11 * 12 * This library is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Library General Public License for more details. 16 * 17 * You should have received a copy of the GNU Library General Public 18 * License along with this library; if not, write to the 19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, 20 * Boston, MA 02110-1301, USA. 21 */ 22 23 #ifndef __GST_AUDIO_AUDIO_H__ 24 #include <gst/audio/audio.h> 25 #endif 26 27 #ifndef __GST_AUDIO_RING_BUFFER_H__ 28 #define __GST_AUDIO_RING_BUFFER_H__ 29 30 G_BEGIN_DECLS 31 32 #define GST_TYPE_AUDIO_RING_BUFFER (gst_audio_ring_buffer_get_type()) 33 #define GST_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBuffer)) 34 #define GST_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBufferClass)) 35 #define GST_AUDIO_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_RING_BUFFER, GstAudioRingBufferClass)) 36 #define GST_AUDIO_RING_BUFFER_CAST(obj) ((GstAudioRingBuffer *)obj) 37 #define GST_IS_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RING_BUFFER)) 38 #define GST_IS_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RING_BUFFER)) 39 40 typedef struct _GstAudioRingBuffer GstAudioRingBuffer; 41 typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass; 42 typedef struct _GstAudioRingBufferSpec GstAudioRingBufferSpec; 43 44 /** 45 * GstAudioRingBufferCallback: 46 * @rbuf: a #GstAudioRingBuffer 47 * @data: (array length=len): target to fill 48 * @len: amount to fill 49 * @user_data: user data 50 * 51 * This function is set with gst_audio_ring_buffer_set_callback() and is 52 * called to fill the memory at @data with @len bytes of samples. 53 */ 54 typedef void (*GstAudioRingBufferCallback) (GstAudioRingBuffer *rbuf, guint8* data, guint len, gpointer user_data); 55 56 /** 57 * GstAudioRingBufferState: 58 * @GST_AUDIO_RING_BUFFER_STATE_STOPPED: The ringbuffer is stopped 59 * @GST_AUDIO_RING_BUFFER_STATE_PAUSED: The ringbuffer is paused 60 * @GST_AUDIO_RING_BUFFER_STATE_STARTED: The ringbuffer is started 61 * @GST_AUDIO_RING_BUFFER_STATE_ERROR: The ringbuffer has encountered an 62 * error after it has been started, e.g. because the device was 63 * disconnected (Since: 1.2) 64 * 65 * The state of the ringbuffer. 66 */ 67 typedef enum { 68 GST_AUDIO_RING_BUFFER_STATE_STOPPED, 69 GST_AUDIO_RING_BUFFER_STATE_PAUSED, 70 GST_AUDIO_RING_BUFFER_STATE_STARTED, 71 GST_AUDIO_RING_BUFFER_STATE_ERROR 72 } GstAudioRingBufferState; 73 74 /** 75 * GstAudioRingBufferFormatType: 76 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: samples in linear or float 77 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: samples in mulaw 78 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: samples in alaw 79 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: samples in ima adpcm 80 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: samples in mpeg audio (but not AAC) format 81 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM: samples in gsm format 82 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958: samples in IEC958 frames (e.g. AC3) 83 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: samples in AC3 format 84 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: samples in EAC3 format 85 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: samples in DTS format 86 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC ADTS format 87 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC ADTS format 88 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: samples in MPEG-2 AAC raw format (Since: 1.12) 89 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: samples in MPEG-4 AAC raw format (Since: 1.12) 90 * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: samples in FLAC format (Since: 1.12) 91 * 92 * The format of the samples in the ringbuffer. 93 */ 94 typedef enum 95 { 96 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW, 97 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW, 98 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW, 99 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM, 100 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG, 101 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM, 102 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958, 103 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3, 104 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3, 105 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS, 106 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC, 107 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC, 108 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW, 109 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW, 110 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC 111 } GstAudioRingBufferFormatType; 112 113 /** 114 * GstAudioRingBufferSpec: 115 * @caps: The caps that generated the Spec. 116 * @type: the sample type 117 * @info: the #GstAudioInfo 118 * @latency_time: the latency in microseconds 119 * @buffer_time: the total buffer size in microseconds 120 * @segsize: the size of one segment in bytes 121 * @segtotal: the total number of segments 122 * @seglatency: number of segments queued in the lower level device, 123 * defaults to segtotal 124 * 125 * The structure containing the format specification of the ringbuffer. 126 */ 127 struct _GstAudioRingBufferSpec 128 { 129 /*< public >*/ 130 /* in */ 131 GstCaps *caps; /* the caps of the buffer */ 132 133 /* in/out */ 134 GstAudioRingBufferFormatType type; 135 GstAudioInfo info; 136 137 138 guint64 latency_time; /* the required/actual latency time, this is the 139 * actual the size of one segment and the 140 * minimum possible latency we can achieve. */ 141 guint64 buffer_time; /* the required/actual time of the buffer, this is 142 * the total size of the buffer and maximum 143 * latency we can compensate for. */ 144 gint segsize; /* size of one buffer segment in bytes, this value 145 * should be chosen to match latency_time as 146 * well as possible. */ 147 gint segtotal; /* total number of segments, this value is the 148 * number of segments of @segsize and should be 149 * chosen so that it matches buffer_time as 150 * close as possible. */ 151 /* ABI added 0.10.20 */ 152 gint seglatency; /* number of segments queued in the lower 153 * level device, defaults to segtotal. */ 154 155 /*< private >*/ 156 gpointer _gst_reserved[GST_PADDING]; 157 }; 158 159 #define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond)) 160 #define GST_AUDIO_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf))) 161 #define GST_AUDIO_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf))) 162 #define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf))) 163 164 /** 165 * GstAudioRingBuffer: 166 * @cond: used to signal start/stop/pause/resume actions 167 * @open: boolean indicating that the ringbuffer is open 168 * @acquired: boolean indicating that the ringbuffer is acquired 169 * @memory: data in the ringbuffer 170 * @size: size of data in the ringbuffer 171 * @spec: format and layout of the ringbuffer data 172 * @samples_per_seg: number of samples in one segment 173 * @empty_seg: pointer to memory holding one segment of silence samples 174 * @state: state of the buffer 175 * @segdone: readpointer in the ringbuffer 176 * @segbase: segment corresponding to segment 0 (unused) 177 * @waiting: is a reader or writer waiting for a free segment 178 * 179 * The ringbuffer base class structure. 180 */ 181 struct _GstAudioRingBuffer { 182 GstObject object; 183 184 /*< public >*/ /* with LOCK */ 185 GCond cond; 186 gboolean open; 187 gboolean acquired; 188 guint8 *memory; 189 gsize size; 190 /*< private >*/ 191 GstClockTime *timestamps; 192 /*< public >*/ /* with LOCK */ 193 GstAudioRingBufferSpec spec; 194 gint samples_per_seg; 195 guint8 *empty_seg; 196 197 /*< public >*/ /* ATOMIC */ 198 gint state; 199 gint segdone; 200 gint segbase; 201 gint waiting; 202 203 /*< private >*/ 204 GstAudioRingBufferCallback callback; 205 gpointer cb_data; 206 207 gboolean need_reorder; 208 /* gst[channel_reorder_map[i]] = device[i] */ 209 gint channel_reorder_map[64]; 210 211 gboolean flushing; 212 /* ATOMIC */ 213 gint may_start; 214 gboolean active; 215 216 GDestroyNotify cb_data_notify; 217 218 /*< private >*/ 219 gpointer _gst_reserved[GST_PADDING - 1]; 220 }; 221 222 /** 223 * GstAudioRingBufferClass: 224 * @parent_class: parent class 225 * @open_device: open the device, don't set any params or allocate anything 226 * @acquire: allocate the resources for the ringbuffer using the given spec 227 * @release: free resources of the ringbuffer 228 * @close_device: close the device 229 * @start: start processing of samples 230 * @pause: pause processing of samples 231 * @resume: resume processing of samples after pause 232 * @stop: stop processing of samples 233 * @delay: get number of frames queued in device 234 * @activate: activate the thread that starts pulling and monitoring the 235 * consumed segments in the device. 236 * @commit: write samples into the ringbuffer 237 * @clear_all: clear the entire ringbuffer. 238 * 239 * The vmethods that subclasses can override to implement the ringbuffer. 240 */ 241 struct _GstAudioRingBufferClass { 242 GstObjectClass parent_class; 243 244 /*< public >*/ 245 gboolean (*open_device) (GstAudioRingBuffer *buf); 246 gboolean (*acquire) (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec); 247 gboolean (*release) (GstAudioRingBuffer *buf); 248 gboolean (*close_device) (GstAudioRingBuffer *buf); 249 250 gboolean (*start) (GstAudioRingBuffer *buf); 251 gboolean (*pause) (GstAudioRingBuffer *buf); 252 gboolean (*resume) (GstAudioRingBuffer *buf); 253 gboolean (*stop) (GstAudioRingBuffer *buf); 254 255 guint (*delay) (GstAudioRingBuffer *buf); 256 257 /* ABI added */ 258 gboolean (*activate) (GstAudioRingBuffer *buf, gboolean active); 259 260 guint (*commit) (GstAudioRingBuffer * buf, guint64 *sample, 261 guint8 * data, gint in_samples, 262 gint out_samples, gint * accum); 263 264 void (*clear_all) (GstAudioRingBuffer * buf); 265 266 /*< private >*/ 267 gpointer _gst_reserved[GST_PADDING]; 268 }; 269 270 GST_AUDIO_API 271 GType gst_audio_ring_buffer_get_type(void); 272 273 /* callback stuff */ 274 275 GST_AUDIO_API 276 void gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf, 277 GstAudioRingBufferCallback cb, 278 gpointer user_data); 279 280 GST_AUDIO_API 281 void gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf, 282 GstAudioRingBufferCallback cb, 283 gpointer user_data, 284 GDestroyNotify notify); 285 286 GST_AUDIO_API 287 gboolean gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec, GstCaps *caps); 288 289 GST_AUDIO_API 290 void gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec); 291 292 GST_AUDIO_API 293 void gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec); 294 295 GST_AUDIO_API 296 gboolean gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf, GstFormat src_fmt, 297 gint64 src_val, GstFormat dest_fmt, 298 gint64 * dest_val); 299 300 /* device state */ 301 302 GST_AUDIO_API 303 gboolean gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf); 304 305 GST_AUDIO_API 306 gboolean gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf); 307 308 GST_AUDIO_API 309 gboolean gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf); 310 311 /* allocate resources */ 312 313 GST_AUDIO_API 314 gboolean gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec); 315 316 GST_AUDIO_API 317 gboolean gst_audio_ring_buffer_release (GstAudioRingBuffer *buf); 318 319 GST_AUDIO_API 320 gboolean gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf); 321 322 /* set the device channel positions */ 323 324 GST_AUDIO_API 325 void gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position); 326 327 /* activating */ 328 329 GST_AUDIO_API 330 gboolean gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf, gboolean active); 331 332 GST_AUDIO_API 333 gboolean gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf); 334 335 /* flushing */ 336 337 GST_AUDIO_API 338 void gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf, gboolean flushing); 339 340 GST_AUDIO_API 341 gboolean gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf); 342 343 /* playback/pause */ 344 345 GST_AUDIO_API 346 gboolean gst_audio_ring_buffer_start (GstAudioRingBuffer *buf); 347 348 GST_AUDIO_API 349 gboolean gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf); 350 351 GST_AUDIO_API 352 gboolean gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf); 353 354 /* get status */ 355 356 GST_AUDIO_API 357 guint gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf); 358 359 GST_AUDIO_API 360 guint64 gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf); 361 362 GST_AUDIO_API 363 void gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf, guint64 sample); 364 365 /* clear all segments */ 366 367 GST_AUDIO_API 368 void gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf); 369 370 /* commit samples */ 371 372 GST_AUDIO_API 373 guint gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 *sample, 374 guint8 * data, gint in_samples, 375 gint out_samples, gint * accum); 376 377 /* read samples */ 378 379 GST_AUDIO_API 380 guint gst_audio_ring_buffer_read (GstAudioRingBuffer *buf, guint64 sample, 381 guint8 *data, guint len, GstClockTime *timestamp); 382 383 /* Set timestamp on buffer */ 384 385 GST_AUDIO_API 386 void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg, GstClockTime 387 timestamp); 388 389 /* mostly protected */ 390 /* not yet implemented 391 gboolean gst_audio_ring_buffer_prepare_write (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len); 392 */ 393 394 GST_AUDIO_API 395 gboolean gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf, gint *segment, 396 guint8 **readptr, gint *len); 397 398 GST_AUDIO_API 399 void gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf, gint segment); 400 401 GST_AUDIO_API 402 void gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf, guint advance); 403 404 GST_AUDIO_API 405 void gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf, gboolean allowed); 406 407 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC 408 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioRingBuffer, gst_object_unref) 409 #endif 410 411 G_END_DECLS 412 413 #endif /* __GST_AUDIO_RING_BUFFER_H__ */ 414