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gstreamer/gst-libs/gst/audio/gstbaseaudiosrc.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosrc.c:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstbaseaudiosrc
* @short_description: Base class for audio sources
* @see_also: #GstAudioSrc, #GstRingBuffer.
*
* This is the base class for audio sources. Subclasses need to implement the
* ::create_ringbuffer vmethod. This base class will then take care of
* reading samples from the ringbuffer, synchronisation and flushing.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include "gstbaseaudiosrc.h"
#include "gst/gst-i18n-plugin.h"
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
#define GST_CAT_DEFAULT gst_base_audio_src_debug
GType
gst_base_audio_src_slave_method_get_type (void)
{
static volatile gsize slave_method_type = 0;
/* FIXME 0.11: nick should be "retimestamp" not "re-timestamp" */
static const GEnumValue slave_method[] = {
{GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
"GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE", "resample"},
{GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
"GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP", "re-timestamp"},
{GST_BASE_AUDIO_SRC_SLAVE_SKEW, "GST_BASE_AUDIO_SRC_SLAVE_SKEW", "skew"},
{GST_BASE_AUDIO_SRC_SLAVE_NONE, "GST_BASE_AUDIO_SRC_SLAVE_NONE", "none"},
{0, NULL, NULL},
};
if (g_once_init_enter (&slave_method_type)) {
GType tmp =
g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method);
g_once_init_leave (&slave_method_type, tmp);
}
return (GType) slave_method_type;
}
#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate))
struct _GstBaseAudioSrcPrivate
{
gboolean provide_clock;
/* the clock slaving algorithm in use */
GstBaseAudioSrcSlaveMethod slave_method;
};
/* BaseAudioSrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_ACTUAL_BUFFER_TIME -1
#define DEFAULT_ACTUAL_LATENCY_TIME -1
#define DEFAULT_PROVIDE_CLOCK TRUE
#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_SKEW
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_ACTUAL_BUFFER_TIME,
PROP_ACTUAL_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
PROP_SLAVE_METHOD,
PROP_LAST
};
static void
_do_init (GType type)
{
GST_DEBUG_CATEGORY_INIT (gst_base_audio_src_debug, "baseaudiosrc", 0,
"baseaudiosrc element");
#ifdef ENABLE_NLS
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
LOCALEDIR);
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif /* ENABLE_NLS */
}
GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_base_audio_src, GstPushSrc,
GST_TYPE_PUSH_SRC, _do_init);
static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_dispose (GObject * object);
static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
element, GstStateChange transition);
static GstClock *gst_base_audio_src_provide_clock (GstElement * elem);
static GstClockTime gst_base_audio_src_get_time (GstClock * clock,
GstBaseAudioSrc * src);
static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc,
guint64 offset, guint length, GstBuffer ** buf);
static gboolean gst_base_audio_src_check_get_range (GstBaseSrc * bsrc);
static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query);
static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
static void
gst_base_audio_src_base_init (gpointer g_class)
{
}
static void
gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
g_type_class_add_private (klass, sizeof (GstBaseAudioSrcPrivate));
gobject_class->set_property = gst_base_audio_src_set_property;
gobject_class->get_property = gst_base_audio_src_get_property;
gobject_class->dispose = gst_base_audio_src_dispose;
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in microseconds", 1,
G_MAXINT64, DEFAULT_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"Audio latency in microseconds", 1,
G_MAXINT64, DEFAULT_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseAudioSrc:actual-buffer-time:
*
* Actual configured size of audio buffer in microseconds.
*
* Since: 0.10.20
**/
g_object_class_install_property (gobject_class, PROP_ACTUAL_BUFFER_TIME,
g_param_spec_int64 ("actual-buffer-time", "Actual Buffer Time",
"Actual configured size of audio buffer in microseconds",
DEFAULT_ACTUAL_BUFFER_TIME, G_MAXINT64, DEFAULT_ACTUAL_BUFFER_TIME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseAudioSrc:actual-latency-time:
*
* Actual configured audio latency in microseconds.
*
* Since: 0.10.20
**/
g_object_class_install_property (gobject_class, PROP_ACTUAL_LATENCY_TIME,
g_param_spec_int64 ("actual-latency-time", "Actual Latency Time",
"Actual configured audio latency in microseconds",
DEFAULT_ACTUAL_LATENCY_TIME, G_MAXINT64, DEFAULT_ACTUAL_LATENCY_TIME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
g_param_spec_boolean ("provide-clock", "Provide Clock",
"Provide a clock to be used as the global pipeline clock",
DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
g_param_spec_enum ("slave-method", "Slave Method",
"Algorithm to use to match the rate of the masterclock",
GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock);
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query);
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
gstbasesrc_class->check_get_range =
GST_DEBUG_FUNCPTR (gst_base_audio_src_check_get_range);
gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
g_type_class_ref (GST_TYPE_RING_BUFFER);
}
static void
gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
GstBaseAudioSrcClass * g_class)
{
baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc);
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
baseaudiosrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
/* reset blocksize we use latency time to calculate a more useful
* value based on negotiated format. */
GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
/* we are always a live source */
gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
/* we operate in time */
gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
}
static void
gst_base_audio_src_dispose (GObject * object)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
GST_OBJECT_LOCK (src);
if (src->clock) {
gst_audio_clock_invalidate (src->clock);
gst_object_unref (src->clock);
src->clock = NULL;
}
if (src->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
}
GST_OBJECT_UNLOCK (src);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstClock *
gst_base_audio_src_provide_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
GstClock *clock;
src = GST_BASE_AUDIO_SRC (elem);
/* we have no ringbuffer (must be NULL state) */
if (src->ringbuffer == NULL)
goto wrong_state;
if (!gst_ring_buffer_is_acquired (src->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (src);
if (!src->priv->provide_clock)
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
GST_OBJECT_UNLOCK (src);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer not acquired");
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (src, "clock provide disabled");
GST_OBJECT_UNLOCK (src);
return NULL;
}
}
static GstClockTime
gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
{
guint64 raw, samples;
guint delay;
GstClockTime result;
if (G_UNLIKELY (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0))
return GST_CLOCK_TIME_NONE;
raw = samples = gst_ring_buffer_samples_done (src->ringbuffer);
/* the number of samples not yet processed, this is still queued in the
* device (not yet read for capture). */
delay = gst_ring_buffer_delay (src->ringbuffer);
samples += delay;
result = gst_util_uint64_scale_int (samples, GST_SECOND,
src->ringbuffer->spec.rate);
GST_DEBUG_OBJECT (src,
"processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT, raw, delay, samples,
GST_TIME_ARGS (result));
return result;
}
static gboolean
gst_base_audio_src_check_get_range (GstBaseSrc * bsrc)
{
/* we allow limited pull base operation of which the details
* will eventually exposed in an as of yet non-existing query.
* Basically pulling can be done on any number of bytes as long
* as the offset is -1 or sequentially increasing. */
return TRUE;
}
/**
* gst_base_audio_src_set_provide_clock:
* @src: a #GstBaseAudioSrc
* @provide: new state
*
* Controls whether @src will provide a clock or not. If @provide is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
*
* Since: 0.10.16
*/
void
gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide)
{
g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
GST_OBJECT_LOCK (src);
src->priv->provide_clock = provide;
GST_OBJECT_UNLOCK (src);
}
/**
* gst_base_audio_src_get_provide_clock:
* @src: a #GstBaseAudioSrc
*
* Queries whether @src will provide a clock or not. See also
* gst_base_audio_src_set_provide_clock.
*
* Returns: %TRUE if @src will provide a clock.
*
* Since: 0.10.16
*/
gboolean
gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src)
{
gboolean result;
g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), FALSE);
GST_OBJECT_LOCK (src);
result = src->priv->provide_clock;
GST_OBJECT_UNLOCK (src);
return result;
}
/**
* gst_base_audio_src_set_slave_method:
* @src: a #GstBaseAudioSrc
* @method: the new slave method
*
* Controls how clock slaving will be performed in @src.
*
* Since: 0.10.20
*/
void
gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src,
GstBaseAudioSrcSlaveMethod method)
{
g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
GST_OBJECT_LOCK (src);
src->priv->slave_method = method;
GST_OBJECT_UNLOCK (src);
}
/**
* gst_base_audio_src_get_slave_method:
* @src: a #GstBaseAudioSrc
*
* Get the current slave method used by @src.
*
* Returns: The current slave method used by @src.
*
* Since: 0.10.20
*/
GstBaseAudioSrcSlaveMethod
gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src)
{
GstBaseAudioSrcSlaveMethod result;
g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1);
GST_OBJECT_LOCK (src);
result = src->priv->slave_method;
GST_OBJECT_UNLOCK (src);
return result;
}
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
src->buffer_time = g_value_get_int64 (value);
break;
case PROP_LATENCY_TIME:
src->latency_time = g_value_get_int64 (value);
break;
case PROP_PROVIDE_CLOCK:
gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
gst_base_audio_src_set_slave_method (src, g_value_get_enum (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, src->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
case PROP_ACTUAL_BUFFER_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.buffer_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_BUFFER_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_ACTUAL_LATENCY_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.latency_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_LATENCY_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
break;
case PROP_SLAVE_METHOD:
g_value_set_enum (value, gst_base_audio_src_get_slave_method (src));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
{
GstStructure *s;
gint width, depth;
s = gst_caps_get_structure (caps, 0);
/* fields for all formats */
gst_structure_fixate_field_nearest_int (s, "rate", 44100);
gst_structure_fixate_field_nearest_int (s, "channels", 2);
gst_structure_fixate_field_nearest_int (s, "width", 16);
/* fields for int */
if (gst_structure_has_field (s, "depth")) {
gst_structure_get_int (s, "width", &width);
/* round width to nearest multiple of 8 for the depth */
depth = GST_ROUND_UP_8 (width);
gst_structure_fixate_field_nearest_int (s, "depth", depth);
}
if (gst_structure_has_field (s, "signed"))
gst_structure_fixate_field_boolean (s, "signed", TRUE);
if (gst_structure_has_field (s, "endianness"))
gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
}
static gboolean
gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstRingBufferSpec *spec;
spec = &src->ringbuffer->spec;
spec->buffer_time = src->buffer_time;
spec->latency_time = src->latency_time;
if (!gst_ring_buffer_parse_caps (spec, caps))
goto parse_error;
/* calculate suggested segsize and segtotal */
spec->segsize =
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
spec->segtotal = spec->buffer_time / spec->latency_time;
GST_DEBUG ("release old ringbuffer");
gst_ring_buffer_release (src->ringbuffer);
gst_ring_buffer_debug_spec_buff (spec);
GST_DEBUG ("acquire new ringbuffer");
if (!gst_ring_buffer_acquire (src->ringbuffer, spec))
goto acquire_error;
/* calculate actual latency and buffer times */
spec->latency_time =
spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample);
spec->buffer_time =
spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
spec->bytes_per_sample);
gst_ring_buffer_debug_spec_buff (spec);
g_object_notify (G_OBJECT (src), "actual-buffer-time");
g_object_notify (G_OBJECT (src), "actual-latency-time");
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
acquire_error:
{
GST_DEBUG ("could not acquire ringbuffer");
return FALSE;
}
}
static void
gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* no need to sync to a clock here, we schedule the samples based
* on our own clock for the moment. */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min_latency, max_latency;
GstRingBufferSpec *spec;
if (G_UNLIKELY (src->ringbuffer == NULL
|| src->ringbuffer->spec.rate == 0))
goto done;
spec = &src->ringbuffer->spec;
/* we have at least 1 segment of latency */
min_latency =
gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
/* we cannot delay more than the buffersize else we lose data */
max_latency =
gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
/* we are always live, the min latency is 1 segment and the max latency is
* the complete buffer of segments. */
gst_query_set_latency (query, TRUE, min_latency, max_latency);
res = TRUE;
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
done:
return res;
}
static gboolean
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
gboolean res;
res = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
GST_DEBUG_OBJECT (bsrc, "flush-start");
gst_ring_buffer_pause (src->ringbuffer);
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
GST_DEBUG_OBJECT (bsrc, "flush-stop");
/* always resync on sample after a flush */
src->next_sample = -1;
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_SEEK:
GST_DEBUG_OBJECT (bsrc, "refuse to seek");
res = FALSE;
break;
default:
GST_DEBUG_OBJECT (bsrc, "dropping event %p", event);
break;
}
return res;
}
/* get the next offset in the ringbuffer for reading samples.
* If the next sample is too far away, this function will position itself to the
* next most recent sample, creating discontinuity */
static guint64
gst_base_audio_src_get_offset (GstBaseAudioSrc * src)
{
guint64 sample;
gint readseg, segdone, segtotal, sps;
gint diff;
/* assume we can append to the previous sample */
sample = src->next_sample;
sps = src->ringbuffer->samples_per_seg;
segtotal = src->ringbuffer->spec.segtotal;
/* get the currently processed segment */
segdone = g_atomic_int_get (&src->ringbuffer->segdone)
- src->ringbuffer->segbase;
if (sample != -1) {
GST_DEBUG_OBJECT (src, "at segment %d and sample %" G_GUINT64_FORMAT,
segdone, sample);
/* figure out the segment and the offset inside the segment where
* the sample should be read from. */
readseg = sample / sps;
/* see how far away it is from the read segment, normally segdone (where new
* data is written in the ringbuffer) is bigger than readseg (where we are
* reading). */
diff = segdone - readseg;
if (diff >= segtotal) {
GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone);
/* sample would be dropped, position to next playable position */
sample = ((guint64) (segdone)) * sps;
}
} else {
/* no previous sample, go to the current position */
GST_DEBUG_OBJECT (src, "first sample, align to current %d", segdone);
sample = ((guint64) (segdone)) * sps;
readseg = segdone;
}
GST_DEBUG_OBJECT (src,
"reading from %d, we are at %d, sample %" G_GUINT64_FORMAT, readseg,
segdone, sample);
return sample;
}
static GstFlowReturn
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
guint samples, total_samples;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
spec = &ringbuffer->spec;
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
bps = spec->bytes_per_sample;
if ((length == 0 && bsrc->blocksize == 0) || length == -1)
/* no length given, use the default segment size */
length = spec->segsize;
else
/* make sure we round down to an integral number of samples */
length -= length % bps;
/* figure out the offset in the ringbuffer */
if (G_UNLIKELY (offset != -1)) {
sample = offset / bps;
/* if a specific offset was given it must be the next sequential
* offset we expect or we fail for now. */
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_base_audio_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);
/* get the number of samples to read */
total_samples = samples = length / bps;
/* FIXME, using a bufferpool would be nice here */
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
GST_DEBUG_OBJECT (src, "wait playing");
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
GST_DEBUG_OBJECT (src, "continue playing");
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
GST_WARNING_OBJECT (src,
"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
("Dropped %" G_GUINT64_FORMAT " samples. This is most likely because "
"downstream can't keep up and is consuming samples too slowly.",
sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
if (!(clock = GST_ELEMENT_CLOCK (src)))
goto no_sync;
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
GstClockTime current_time;
guint64 running_time_sample;
gint running_time_segment;
gint last_read_segment;
gint segment_skew;
gint sps;
gint segments_written;
gint last_written_segment;
/* get the amount of segments written from the device by now */
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
last_written_segment = segments_written - ringbuffer->segbase - 1;
/* samples per segment */
sps = ringbuffer->samples_per_seg;
/* get the current time */
current_time = gst_clock_get_time (clock);
/* get the basetime */
base_time = GST_ELEMENT_CAST (src)->base_time;
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
last_read_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime (last written to) */
segment_skew = running_time_segment - last_written_segment;
GST_DEBUG_OBJECT (bsrc,
"\n running_time = %"
GST_TIME_FORMAT
"\n timestamp = %"
GST_TIME_FORMAT
"\n running_time_segment = %d"
"\n last_written_segment = %d"
"\n segment_skew (running time segment - last_written_segment) = %d"
"\n last_read_segment = %d",
GST_TIME_ARGS (running_time), GST_TIME_ARGS (timestamp),
running_time_segment, last_written_segment, segment_skew,
last_read_segment);
/* Resync the ringbuffer if:
*
* 1. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
*
* 2. If this is our first buffer.
* We know that we should catch up to running_time
* the first time we are ran.
*/
if ((segment_skew >= ringbuffer->spec.segtotal) ||
(last_read_segment == 0)) {
gint new_read_segment;
gint segment_diff;
guint64 new_sample;
/* the difference between running_time and the last written segment */
segment_diff = running_time_segment - last_written_segment;
/* advance the ringbuffer */
gst_ring_buffer_advance (ringbuffer, segment_diff);
/* we move the new read segment to the last known written segment */
new_read_segment =
g_atomic_int_get (&ringbuffer->segdone) - ringbuffer->segbase;
/* we calculate the new sample value */
new_sample = ((guint64) new_read_segment) * sps;
/* and get the relative time to this -> our new timestamp */
timestamp =
gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
/* we update the next sample accordingly */
src->next_sample = new_sample + samples;
GST_DEBUG_OBJECT (bsrc,
"Timeshifted the ringbuffer with %d segments: "
"Updating the timestamp to %" GST_TIME_FORMAT ", "
"and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
GST_TIME_ARGS (timestamp), src->next_sample);
}
break;
}
case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
if (GST_CLOCK_DIFF (timestamp, base_time) < 0)
timestamp -= base_time;
else
timestamp = 0;
/* subtract latency */
latency =
gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
}
case GST_BASE_AUDIO_SRC_SLAVE_NONE:
break;
}
} else {
GstClockTime base_time;
/* to get the timestamp against the clock we also need to add our offset */
timestamp = gst_audio_clock_adjust (clock, timestamp);
/* we are not slaved, subtract base_time */
base_time = GST_ELEMENT_CAST (src)->base_time;
if (GST_CLOCK_DIFF (timestamp, base_time) < 0) {
timestamp -= base_time;
GST_LOG_OBJECT (src,
"buffer timestamp %" GST_TIME_FORMAT " (base_time %" GST_TIME_FORMAT
")", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (base_time));
} else {
GST_LOG_OBJECT (src,
"buffer timestamp 0, ts %" GST_TIME_FORMAT " <= base_time %"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (base_time));
timestamp = 0;
}
}
no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
wrong_offset:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
(NULL), ("resource can only be operated on sequentially but offset %"
G_GUINT64_FORMAT " was given", offset));
return GST_FLOW_ERROR;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
/**
* gst_base_audio_src_create_ringbuffer:
* @src: a #GstBaseAudioSrc.
*
* Create and return the #GstRingBuffer for @src. This function will call the
* ::create_ringbuffer vmethod and will set @src as the parent of the returned
* buffer (see gst_object_set_parent()).
*
* Returns: The new ringbuffer of @src.
*/
GstRingBuffer *
gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstBaseAudioSrcClass *bclass;
GstRingBuffer *buffer = NULL;
bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (src);
if (G_LIKELY (buffer))
gst_object_set_parent (GST_OBJECT_CAST (buffer), GST_OBJECT_CAST (src));
return buffer;
}
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (src, "READY->PAUSED");
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
src->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
/* Only post clock-lost messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_release (src->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
GST_OBJECT_UNLOCK (src);
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
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