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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h
Sebastian Dröge 8ea355e52c audioringbuffer: Avoid overflows of segment done counter 
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.

As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.

All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.

This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
2024-05-16 06:52:58 +00:00

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/* 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__
#include <gst/audio/gstdsdformat.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;
typedef struct _GstAudioRingBufferPrivate GstAudioRingBufferPrivate;
/**
* 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)
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD: samples in DSD format (Since: 1.24)
*
* 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,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD
} 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
* @dsd_format: the #GstDsdFormat (Since: 1.24)
*
* The structure containing the format specification of the ringbuffer.
*
* When @type is GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD, the @dsd_format
* is valid (otherwise it is unused). Also, when DSD is the sample type,
* only the rate, channels, position, and bpf fields in @info are populated.
*/
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. */
/* Union preserves padded struct size for backwards compat
* Consumer code should use the accessor macros for fields */
union {
struct { /* < skip > */
GstDsdFormat dsd_format;
} abi;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
} ABI;
};
#define GST_AUDIO_RING_BUFFER_SPEC_FORMAT_TYPE(spec) ((spec)->type)
#define GST_AUDIO_RING_BUFFER_SPEC_INFO(spec) ((spec)->info)
#define GST_AUDIO_RING_BUFFER_SPEC_LATENCY_TIME(spec) ((spec)->latency_time)
#define GST_AUDIO_RING_BUFFER_SPEC_BUFFER_TIME(spec) ((spec)->buffer_time)
#define GST_AUDIO_RING_BUFFER_SPEC_SEGSIZE(spec) ((spec)->segsize)
#define GST_AUDIO_RING_BUFFER_SPEC_SEGTOTAL(spec) ((spec)->segtotal)
#define GST_AUDIO_RING_BUFFER_SPEC_SEGLATENCY(spec) ((spec)->seglatency)
#define GST_AUDIO_RING_BUFFER_SPEC_DSD_FORMAT(spec) ((spec)->ABI.abi.dsd_format)
#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;
/*< private >*/
GstClockTime *timestamps;
/*< public >*/ /* with LOCK */
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 >*/
GstAudioRingBufferPrivate *priv;
gpointer _gst_reserved[GST_PADDING - 2];
};
/**
* 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: Optional.
* Clear the entire ringbuffer.
* Subclasses should chain up to the parent implementation to
* invoke the default handler.
*
* 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);
GST_AUDIO_API
void gst_audio_ring_buffer_set_errored (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);
GST_AUDIO_API
void gst_audio_ring_buffer_set_segdone (GstAudioRingBuffer *buf, guint64 segdone);
GST_AUDIO_API
guint64 gst_audio_ring_buffer_get_segdone (GstAudioRingBuffer *buf);
GST_AUDIO_API
guint64 gst_audio_ring_buffer_get_segbase (GstAudioRingBuffer *buf);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioRingBuffer, gst_object_unref)
G_END_DECLS
#endif /* __GST_AUDIO_RING_BUFFER_H__ */
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