| /* GStreamer |
| * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
| * 2005 Wim Taymans <wim@fluendo.com> |
| * |
| * gstaudioringbuffer.h: |
| * |
| * 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. |
| */ |
| |
| #ifndef __GST_AUDIO_AUDIO_H__ |
| #include <gst/audio/audio.h> |
| #endif |
| |
| #ifndef __GST_AUDIO_RING_BUFFER_H__ |
| #define __GST_AUDIO_RING_BUFFER_H__ |
| |
| G_BEGIN_DECLS |
| |
| #define GST_TYPE_AUDIO_RING_BUFFER (gst_audio_ring_buffer_get_type()) |
| #define GST_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBuffer)) |
| #define GST_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBufferClass)) |
| #define GST_AUDIO_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_RING_BUFFER, GstAudioRingBufferClass)) |
| #define GST_AUDIO_RING_BUFFER_CAST(obj) ((GstAudioRingBuffer *)obj) |
| #define GST_IS_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RING_BUFFER)) |
| #define GST_IS_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RING_BUFFER)) |
| |
| typedef struct _GstAudioRingBuffer GstAudioRingBuffer; |
| typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass; |
| typedef struct _GstAudioRingBufferSpec GstAudioRingBufferSpec; |
| |
| /** |
| * GstAudioRingBufferCallback: |
| * @rbuf: a #GstAudioRingBuffer |
| * @data: (array length=len): target to fill |
| * @len: amount to fill |
| * @user_data: user data |
| * |
| * This function is set with gst_audio_ring_buffer_set_callback() and is |
| * called to fill the memory at @data with @len bytes of samples. |
| */ |
| typedef void (*GstAudioRingBufferCallback) (GstAudioRingBuffer *rbuf, guint8* data, guint len, gpointer user_data); |
| |
| /** |
| * GstAudioRingBufferState: |
| * @GST_AUDIO_RING_BUFFER_STATE_STOPPED: The ringbuffer is stopped |
| * @GST_AUDIO_RING_BUFFER_STATE_PAUSED: The ringbuffer is paused |
| * @GST_AUDIO_RING_BUFFER_STATE_STARTED: The ringbuffer is started |
| * @GST_AUDIO_RING_BUFFER_STATE_ERROR: The ringbuffer has encountered an |
| * error after it has been started, e.g. because the device was |
| * disconnected (Since 1.2) |
| * |
| * The state of the ringbuffer. |
| */ |
| typedef enum { |
| GST_AUDIO_RING_BUFFER_STATE_STOPPED, |
| GST_AUDIO_RING_BUFFER_STATE_PAUSED, |
| GST_AUDIO_RING_BUFFER_STATE_STARTED, |
| GST_AUDIO_RING_BUFFER_STATE_ERROR |
| } GstAudioRingBufferState; |
| |
| /** |
| * GstAudioRingBufferFormatType: |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: samples in linear or float |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: samples in mulaw |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: samples in alaw |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: samples in ima adpcm |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: samples in mpeg audio (but not AAC) format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM: samples in gsm format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958: samples in IEC958 frames (e.g. AC3) |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: samples in AC3 format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: samples in EAC3 format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: samples in DTS format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC ADTS format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC ADTS format |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: samples in MPEG-2 AAC raw format (Since 1.12) |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: samples in MPEG-4 AAC raw format (Since 1.12) |
| * @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: samples in FLAC format (Since 1.12) |
| * |
| * The format of the samples in the ringbuffer. |
| */ |
| typedef enum |
| { |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW, |
| GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC |
| } GstAudioRingBufferFormatType; |
| |
| /** |
| * GstAudioRingBufferSpec: |
| * @caps: The caps that generated the Spec. |
| * @type: the sample type |
| * @info: the #GstAudioInfo |
| * @latency_time: the latency in microseconds |
| * @buffer_time: the total buffer size in microseconds |
| * @segsize: the size of one segment in bytes |
| * @segtotal: the total number of segments |
| * @seglatency: number of segments queued in the lower level device, |
| * defaults to segtotal |
| * |
| * The structure containing the format specification of the ringbuffer. |
| */ |
| struct _GstAudioRingBufferSpec |
| { |
| /*< public >*/ |
| /* in */ |
| GstCaps *caps; /* the caps of the buffer */ |
| |
| /* in/out */ |
| GstAudioRingBufferFormatType type; |
| GstAudioInfo info; |
| |
| |
| guint64 latency_time; /* the required/actual latency time, this is the |
| * actual the size of one segment and the |
| * minimum possible latency we can achieve. */ |
| guint64 buffer_time; /* the required/actual time of the buffer, this is |
| * the total size of the buffer and maximum |
| * latency we can compensate for. */ |
| gint segsize; /* size of one buffer segment in bytes, this value |
| * should be chosen to match latency_time as |
| * well as possible. */ |
| gint segtotal; /* total number of segments, this value is the |
| * number of segments of @segsize and should be |
| * chosen so that it matches buffer_time as |
| * close as possible. */ |
| /* ABI added 0.10.20 */ |
| gint seglatency; /* number of segments queued in the lower |
| * level device, defaults to segtotal. */ |
| |
| /*< private >*/ |
| gpointer _gst_reserved[GST_PADDING]; |
| }; |
| |
| #define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond)) |
| #define GST_AUDIO_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf))) |
| #define GST_AUDIO_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf))) |
| #define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf))) |
| |
| /** |
| * GstAudioRingBuffer: |
| * @cond: used to signal start/stop/pause/resume actions |
| * @open: boolean indicating that the ringbuffer is open |
| * @acquired: boolean indicating that the ringbuffer is acquired |
| * @memory: data in the ringbuffer |
| * @size: size of data in the ringbuffer |
| * @spec: format and layout of the ringbuffer data |
| * @samples_per_seg: number of samples in one segment |
| * @empty_seg: pointer to memory holding one segment of silence samples |
| * @state: state of the buffer |
| * @segdone: readpointer in the ringbuffer |
| * @segbase: segment corresponding to segment 0 (unused) |
| * @waiting: is a reader or writer waiting for a free segment |
| * |
| * The ringbuffer base class structure. |
| */ |
| struct _GstAudioRingBuffer { |
| GstObject object; |
| |
| /*< public >*/ /* with LOCK */ |
| GCond cond; |
| gboolean open; |
| gboolean acquired; |
| guint8 *memory; |
| gsize size; |
| GstClockTime *timestamps; |
| GstAudioRingBufferSpec spec; |
| gint samples_per_seg; |
| guint8 *empty_seg; |
| |
| /*< public >*/ /* ATOMIC */ |
| gint state; |
| gint segdone; |
| gint segbase; |
| gint waiting; |
| |
| /*< private >*/ |
| GstAudioRingBufferCallback callback; |
| gpointer cb_data; |
| |
| gboolean need_reorder; |
| /* gst[channel_reorder_map[i]] = device[i] */ |
| gint channel_reorder_map[64]; |
| |
| gboolean flushing; |
| /* ATOMIC */ |
| gint may_start; |
| gboolean active; |
| |
| GDestroyNotify cb_data_notify; |
| |
| /*< private >*/ |
| gpointer _gst_reserved[GST_PADDING - 1]; |
| }; |
| |
| /** |
| * GstAudioRingBufferClass: |
| * @parent_class: parent class |
| * @open_device: open the device, don't set any params or allocate anything |
| * @acquire: allocate the resources for the ringbuffer using the given spec |
| * @release: free resources of the ringbuffer |
| * @close_device: close the device |
| * @start: start processing of samples |
| * @pause: pause processing of samples |
| * @resume: resume processing of samples after pause |
| * @stop: stop processing of samples |
| * @delay: get number of frames queued in device |
| * @activate: activate the thread that starts pulling and monitoring the |
| * consumed segments in the device. |
| * @commit: write samples into the ringbuffer |
| * @clear_all: clear the entire ringbuffer. |
| * |
| * The vmethods that subclasses can override to implement the ringbuffer. |
| */ |
| struct _GstAudioRingBufferClass { |
| GstObjectClass parent_class; |
| |
| /*< public >*/ |
| gboolean (*open_device) (GstAudioRingBuffer *buf); |
| gboolean (*acquire) (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec); |
| gboolean (*release) (GstAudioRingBuffer *buf); |
| gboolean (*close_device) (GstAudioRingBuffer *buf); |
| |
| gboolean (*start) (GstAudioRingBuffer *buf); |
| gboolean (*pause) (GstAudioRingBuffer *buf); |
| gboolean (*resume) (GstAudioRingBuffer *buf); |
| gboolean (*stop) (GstAudioRingBuffer *buf); |
| |
| guint (*delay) (GstAudioRingBuffer *buf); |
| |
| /* ABI added */ |
| gboolean (*activate) (GstAudioRingBuffer *buf, gboolean active); |
| |
| guint (*commit) (GstAudioRingBuffer * buf, guint64 *sample, |
| guint8 * data, gint in_samples, |
| gint out_samples, gint * accum); |
| |
| void (*clear_all) (GstAudioRingBuffer * buf); |
| |
| /*< private >*/ |
| gpointer _gst_reserved[GST_PADDING]; |
| }; |
| |
| GST_AUDIO_API |
| GType gst_audio_ring_buffer_get_type(void); |
| |
| /* callback stuff */ |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf, |
| GstAudioRingBufferCallback cb, |
| gpointer user_data); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf, |
| GstAudioRingBufferCallback cb, |
| gpointer user_data, |
| GDestroyNotify notify); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec, GstCaps *caps); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf, GstFormat src_fmt, |
| gint64 src_val, GstFormat dest_fmt, |
| gint64 * dest_val); |
| |
| /* device state */ |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf); |
| |
| /* allocate resources */ |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_release (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf); |
| |
| /* set the device channel positions */ |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position); |
| |
| /* activating */ |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf, gboolean active); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf); |
| |
| /* flushing */ |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf, gboolean flushing); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf); |
| |
| /* playback/pause */ |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_start (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf); |
| |
| /* get status */ |
| |
| GST_AUDIO_API |
| guint gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| guint64 gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf, guint64 sample); |
| |
| /* clear all segments */ |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf); |
| |
| /* commit samples */ |
| |
| GST_AUDIO_API |
| guint gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 *sample, |
| guint8 * data, gint in_samples, |
| gint out_samples, gint * accum); |
| |
| /* read samples */ |
| |
| GST_AUDIO_API |
| guint gst_audio_ring_buffer_read (GstAudioRingBuffer *buf, guint64 sample, |
| guint8 *data, guint len, GstClockTime *timestamp); |
| |
| /* Set timestamp on buffer */ |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg, GstClockTime |
| timestamp); |
| |
| /* mostly protected */ |
| /* not yet implemented |
| gboolean gst_audio_ring_buffer_prepare_write (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len); |
| */ |
| |
| GST_AUDIO_API |
| gboolean gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf, gint *segment, |
| guint8 **readptr, gint *len); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf, gint segment); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf, guint advance); |
| |
| GST_AUDIO_API |
| void gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf, gboolean allowed); |
| |
| #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC |
| G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioRingBuffer, gst_object_unref) |
| #endif |
| |
| G_END_DECLS |
| |
| #endif /* __GST_AUDIO_RING_BUFFER_H__ */ |
