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gstreamer/gst-libs/gst/audio/gstbaseaudiosink.c
Tim-Philipp Müller b52c5819fb Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
2011-11-01 00:34:28 +00:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosink.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:gstbaseaudiosink
* @short_description: Base class for audio sinks
* @see_also: #GstAudioSink, #GstRingBuffer.
*
* This is the base class for audio sinks. Subclasses need to implement the
* ::create_ringbuffer vmethod. This base class will then take care of
* writing samples to the ringbuffer, synchronisation, clipping and flushing.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
#include <string.h>
#include "gstbaseaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug);
#define GST_CAT_DEFAULT gst_base_audio_sink_debug
#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate))
struct _GstBaseAudioSinkPrivate
{
/* upstream latency */
GstClockTime us_latency;
/* the clock slaving algorithm in use */
GstBaseAudioSinkSlaveMethod slave_method;
/* running average of clock skew */
GstClockTimeDiff avg_skew;
/* the number of samples we aligned last time */
gint64 last_align;
gboolean sync_latency;
GstClockTime eos_time;
/* number of microseconds we alow timestamps or clock slaving to drift
* before resyncing */
guint64 drift_tolerance;
};
/* BaseAudioSink signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
/* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_PROVIDE_CLOCK TRUE
#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SINK_SLAVE_SKEW
/* FIXME, enable pull mode when clock slaving and trick modes are figured out */
#define DEFAULT_CAN_ACTIVATE_PULL FALSE
/* when timestamps or clock slaving drift for more than 40ms we resync. This is
* a reasonable default */
#define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND)
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
PROP_SLAVE_METHOD,
PROP_CAN_ACTIVATE_PULL,
PROP_DRIFT_TOLERANCE,
PROP_LAST
};
GType
gst_base_audio_sink_slave_method_get_type (void)
{
static volatile gsize slave_method_type = 0;
static const GEnumValue slave_method[] = {
{GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE",
"resample"},
{GST_BASE_AUDIO_SINK_SLAVE_SKEW, "GST_BASE_AUDIO_SINK_SLAVE_SKEW", "skew"},
{GST_BASE_AUDIO_SINK_SLAVE_NONE, "GST_BASE_AUDIO_SINK_SLAVE_NONE", "none"},
{0, NULL, NULL},
};
if (g_once_init_enter (&slave_method_type)) {
GType tmp =
g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
g_once_init_leave (&slave_method_type, tmp);
}
return (GType) slave_method_type;
}
#define _do_init \
GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
#define gst_base_audio_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSink, gst_base_audio_sink,
GST_TYPE_BASE_SINK, _do_init);
static void gst_base_audio_sink_dispose (GObject * object);
static void gst_base_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
#if 0
static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
basesink);
#endif
static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
gboolean active);
static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
query);
static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
GstBaseAudioSink * sink);
static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
guint len, gpointer user_data);
static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink,
GstBuffer * buffer);
static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink,
GstBuffer * buffer);
static gboolean gst_base_audio_sink_event (GstBaseSink * bsink,
GstEvent * event);
static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
GstCaps * caps);
static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
static gboolean gst_base_audio_sink_query_pad (GstBaseSink * bsink,
GstQuery * query);
/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
static void
gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate));
gobject_class->set_property = gst_base_audio_sink_set_property;
gobject_class->get_property = gst_base_audio_sink_get_property;
gobject_class->dispose = gst_base_audio_sink_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));
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_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
"Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseAudioSink:drift-tolerance
*
* Controls the amount of time in milliseconds that timestamps or clocks are allowed
* to drift before resynchronisation happens.
*
* Since: 0.10.26
*/
g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
"Tolerance for timestamp and clock drift in microseconds", 1,
G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render);
gstbasesink_class->get_times =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate);
#if 0
gstbasesink_class->async_play =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
#endif
gstbasesink_class->activate_pull =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull);
/* 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_sink_init (GstBaseAudioSink * baseaudiosink)
{
GstBaseSink *basesink;
baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD;
baseaudiosink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
basesink = GST_BASE_SINK_CAST (baseaudiosink);
basesink->can_activate_push = TRUE;
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
gst_base_sink_set_last_buffer_enabled (basesink, FALSE);
}
static void
gst_base_audio_sink_dispose (GObject * object)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (object);
if (sink->provided_clock) {
gst_audio_clock_invalidate (sink->provided_clock);
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
}
if (sink->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstClock *
gst_base_audio_sink_provide_clock (GstElement * elem)
{
GstBaseAudioSink *sink;
GstClock *clock;
sink = GST_BASE_AUDIO_SINK (elem);
/* we have no ringbuffer (must be NULL state) */
if (sink->ringbuffer == NULL)
goto wrong_state;
if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (sink);
if (!sink->provide_clock)
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
GST_OBJECT_UNLOCK (sink);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (sink, "clock provide disabled");
GST_OBJECT_UNLOCK (sink);
return NULL;
}
}
static gboolean
gst_base_audio_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
{
gboolean res = FALSE;
GstBaseAudioSink *basesink;
basesink = GST_BASE_AUDIO_SINK (bsink);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_LOG_OBJECT (basesink, "query convert");
if (basesink->ringbuffer) {
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res = gst_ring_buffer_convert (basesink->ringbuffer, src_fmt, src_val,
dest_fmt, &dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
}
break;
}
default:
res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
break;
}
return res;
}
static gboolean
gst_base_audio_sink_query (GstElement * element, GstQuery * query)
{
gboolean res = FALSE;
GstBaseAudioSink *basesink;
basesink = GST_BASE_AUDIO_SINK (element);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (basesink, "latency query");
/* ask parent first, it will do an upstream query for us. */
if ((res =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
&us_live, &min_l, &max_l))) {
GstClockTime base_latency, min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
GstRingBufferSpec *spec;
GST_OBJECT_LOCK (basesink);
if (!basesink->ringbuffer || !basesink->ringbuffer->spec.info.rate) {
GST_OBJECT_UNLOCK (basesink);
GST_DEBUG_OBJECT (basesink,
"we are not yet negotiated, can't report latency yet");
res = FALSE;
goto done;
}
spec = &basesink->ringbuffer->spec;
basesink->priv->us_latency = min_l;
base_latency =
gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
GST_SECOND, spec->info.rate * spec->info.bpf);
GST_OBJECT_UNLOCK (basesink);
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = base_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
GST_DEBUG_OBJECT (basesink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (basesink,
"peer max %" GST_TIME_FORMAT ", our max latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
GST_TIME_ARGS (max_latency));
} else {
GST_DEBUG_OBJECT (basesink,
"peer or we are not live, don't care about latency");
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_LOG_OBJECT (basesink, "query convert");
if (basesink->ringbuffer) {
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res = gst_ring_buffer_convert (basesink->ringbuffer, src_fmt, src_val,
dest_fmt, &dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
}
break;
}
default:
res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
break;
}
done:
return res;
}
static GstClockTime
gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
{
guint64 raw, samples;
guint delay;
GstClockTime result;
if (sink->ringbuffer == NULL || sink->ringbuffer->spec.info.rate == 0)
return GST_CLOCK_TIME_NONE;
/* our processed samples are always increasing */
raw = samples = gst_ring_buffer_samples_done (sink->ringbuffer);
/* the number of samples not yet processed, this is still queued in the
* device (not played for playback). */
delay = gst_ring_buffer_delay (sink->ringbuffer);
if (G_LIKELY (samples >= delay))
samples -= delay;
else
samples = 0;
result = gst_util_uint64_scale_int (samples, GST_SECOND,
sink->ringbuffer->spec.info.rate);
GST_DEBUG_OBJECT (sink,
"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;
}
/**
* gst_base_audio_sink_set_provide_clock:
* @sink: a #GstBaseAudioSink
* @provide: new state
*
* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
*
* Since: 0.10.16
*/
void
gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink,
gboolean provide)
{
g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->provide_clock = provide;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_audio_sink_get_provide_clock:
* @sink: a #GstBaseAudioSink
*
* Queries whether @sink will provide a clock or not. See also
* gst_base_audio_sink_set_provide_clock.
*
* Returns: %TRUE if @sink will provide a clock.
*
* Since: 0.10.16
*/
gboolean
gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink)
{
gboolean result;
g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE);
GST_OBJECT_LOCK (sink);
result = sink->provide_clock;
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
* gst_base_audio_sink_set_slave_method:
* @sink: a #GstBaseAudioSink
* @method: the new slave method
*
* Controls how clock slaving will be performed in @sink.
*
* Since: 0.10.16
*/
void
gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink,
GstBaseAudioSinkSlaveMethod method)
{
g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->slave_method = method;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_audio_sink_get_slave_method:
* @sink: a #GstBaseAudioSink
*
* Get the current slave method used by @sink.
*
* Returns: The current slave method used by @sink.
*
* Since: 0.10.16
*/
GstBaseAudioSinkSlaveMethod
gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink)
{
GstBaseAudioSinkSlaveMethod result;
g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->slave_method;
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
* gst_base_audio_sink_set_drift_tolerance:
* @sink: a #GstBaseAudioSink
* @drift_tolerance: the new drift tolerance in microseconds
*
* Controls the sink's drift tolerance.
*
* Since: 0.10.31
*/
void
gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink,
gint64 drift_tolerance)
{
g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->drift_tolerance = drift_tolerance;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_audio_sink_get_drift_tolerance
* @sink: a #GstBaseAudioSink
*
* Get the current drift tolerance, in microseconds, used by @sink.
*
* Returns: The current drift tolerance used by @sink.
*
* Since: 0.10.31
*/
gint64
gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink)
{
gint64 result;
g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->drift_tolerance;
GST_OBJECT_UNLOCK (sink);
return result;
}
static void
gst_base_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
sink->buffer_time = g_value_get_int64 (value);
break;
case PROP_LATENCY_TIME:
sink->latency_time = g_value_get_int64 (value);
break;
case PROP_PROVIDE_CLOCK:
gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value));
break;
case PROP_CAN_ACTIVATE_PULL:
GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
break;
case PROP_DRIFT_TOLERANCE:
gst_base_audio_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, sink->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, sink->latency_time);
break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink));
break;
case PROP_SLAVE_METHOD:
g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink));
break;
case PROP_CAN_ACTIVATE_PULL:
g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
break;
case PROP_DRIFT_TOLERANCE:
g_value_set_int64 (value, gst_base_audio_sink_get_drift_tolerance (sink));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
{
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
GstRingBufferSpec *spec;
GstClockTime now;
GstClockTime crate_num, crate_denom;
if (!sink->ringbuffer)
return FALSE;
spec = &sink->ringbuffer->spec;
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
/* get current time, updates the last_time. When the subclass has a clock that
* restarts from 0 when a new format is negotiated, it will call
* gst_audio_clock_reset() which will use this last_time to create an offset
* so that time from the clock keeps on increasing monotonically. */
now = gst_clock_get_time (sink->provided_clock);
GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
/* release old ringbuffer */
gst_ring_buffer_pause (sink->ringbuffer);
gst_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_ring_buffer_release (sink->ringbuffer);
GST_DEBUG_OBJECT (sink, "parse caps");
spec->buffer_time = sink->buffer_time;
spec->latency_time = sink->latency_time;
/* parse new caps */
if (!gst_ring_buffer_parse_caps (spec, caps))
goto parse_error;
gst_ring_buffer_debug_spec_buff (spec);
GST_DEBUG_OBJECT (sink, "acquire ringbuffer");
if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
goto acquire_error;
if (bsink->pad_mode == GST_PAD_ACTIVATE_PUSH) {
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
gst_ring_buffer_activate (sink->ringbuffer, TRUE);
}
/* due to possible changes in the spec file we should recalibrate the clock */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
&crate_num, &crate_denom);
gst_clock_set_calibration (sink->provided_clock,
gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
crate_denom);
/* calculate actual latency and buffer times.
* FIXME: In 0.11, store the latency_time internally in ns */
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->info.rate * spec->info.bpf);
spec->buffer_time = spec->segtotal * spec->latency_time;
gst_ring_buffer_debug_spec_buff (spec);
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG_OBJECT (sink, "could not parse caps");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
(NULL), ("cannot parse audio format."));
return FALSE;
}
acquire_error:
{
GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
return FALSE;
}
}
static void
gst_base_audio_sink_fixate (GstBaseSink * bsink, 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 void
gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* our clock sync is a bit too much for the base class to handle so
* we implement it ourselves. */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
/* This waits for the drain to happen and can be canceled */
static gboolean
gst_base_audio_sink_drain (GstBaseAudioSink * sink)
{
if (!sink->ringbuffer)
return TRUE;
if (!sink->ringbuffer->spec.info.rate)
return TRUE;
/* if PLAYING is interrupted,
* arrange to have clock running when going to PLAYING again */
g_atomic_int_set (&sink->eos_rendering, 1);
/* need to start playback before we can drain, but only when
* we have successfully negotiated a format and thus acquired the
* ringbuffer. */
if (gst_ring_buffer_is_acquired (sink->ringbuffer))
gst_ring_buffer_start (sink->ringbuffer);
if (sink->priv->eos_time != -1) {
GST_DEBUG_OBJECT (sink,
"last sample time %" GST_TIME_FORMAT,
GST_TIME_ARGS (sink->priv->eos_time));
/* wait for the EOS time to be reached, this is the time when the last
* sample is played. */
gst_base_sink_wait_eos (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
GST_DEBUG_OBJECT (sink, "drained audio");
}
g_atomic_int_set (&sink->eos_rendering, 0);
return TRUE;
}
static gboolean
gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
{
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
if (sink->ringbuffer)
gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
if (sink->ringbuffer)
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
break;
case GST_EVENT_EOS:
/* now wait till we played everything */
gst_base_audio_sink_drain (sink);
break;
default:
break;
}
return TRUE;
}
static GstFlowReturn
gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
{
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
/* we don't really do anything when prerolling. We could make a
* property to play this buffer to have some sort of scrubbing
* support. */
return GST_FLOW_OK;
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static guint64
gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
{
guint64 sample;
gint writeseg, segdone, sps;
gint diff;
/* assume we can append to the previous sample */
sample = sink->next_sample;
/* no previous sample, try to insert at position 0 */
if (sample == -1)
sample = 0;
sps = sink->ringbuffer->samples_per_seg;
/* figure out the segment and the offset inside the segment where
* the sample should be written. */
writeseg = sample / sps;
/* get the currently processed segment */
segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
- sink->ringbuffer->segbase;
/* see how far away it is from the write segment */
diff = writeseg - segdone;
if (diff < 0) {
/* sample would be dropped, position to next playable position */
sample = (segdone + 1) * sps;
}
return sample;
}
static GstClockTime
clock_convert_external (GstClockTime external, GstClockTime cinternal,
GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom)
{
/* adjust for rate and speed */
if (external >= cexternal) {
external =
gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
external += cinternal;
} else {
external =
gst_util_uint64_scale (cexternal - external, crate_denom, crate_num);
if (cinternal > external)
external = cinternal - external;
else
external = 0;
}
return external;
}
/* algorithm to calculate sample positions that will result in resampling to
* match the clock rate of the master */
static void
gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal;
GstClockTime crate_num, crate_denom;
/* FIXME, we can sample and add observations here or use the timeouts on the
* clock. No idea which one is better or more stable. The timeout seems more
* arbitrary but this one seems more demanding and does not work when there is
* no data comming in to the sink. */
#if 0
GstClockTime etime, itime;
gdouble r_squared;
/* sample clocks and figure out clock skew */
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
itime = gst_audio_clock_get_time (sink->provided_clock);
/* add new observation */
gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
#endif
/* get calibration parameters to compensate for speed and offset differences
* when we are slaved */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
crate_denom, gst_guint64_to_gdouble (crate_num) /
gst_guint64_to_gdouble (crate_denom));
if (crate_num == 0)
crate_denom = crate_num = 1;
/* bring external time to internal time */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
GST_DEBUG_OBJECT (sink,
"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
*srender_start = render_start;
*srender_stop = render_stop;
}
/* algorithm to calculate sample positions that will result in changing the
* playout pointer to match the clock rate of the master */
static void
gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal, crate_num, crate_denom;
GstClockTime etime, itime;
GstClockTimeDiff skew, mdrift, mdrift2;
gint driftsamples;
gint64 last_align;
/* get calibration parameters to compensate for offsets */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
/* sample clocks and figure out clock skew */
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
itime = gst_audio_clock_get_time (sink->provided_clock);
itime = gst_audio_clock_adjust (sink->provided_clock, itime);
GST_DEBUG_OBJECT (sink,
"internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
" cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
/* make sure we never go below 0 */
etime = etime > cexternal ? etime - cexternal : 0;
itime = itime > cinternal ? itime - cinternal : 0;
/* do itime - etime.
* positive value means external clock goes slower
* negative value means external clock goes faster */
skew = GST_CLOCK_DIFF (etime, itime);
if (sink->priv->avg_skew == -1) {
/* first observation */
sink->priv->avg_skew = skew;
} else {
/* next observations use a moving average */
sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
}
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
/* the max drift we allow */
mdrift = sink->priv->drift_tolerance * 1000;
mdrift2 = mdrift / 2;
/* adjust playout pointer based on skew */
if (sink->priv->avg_skew > mdrift2) {
/* master is running slower, move internal time forward */
GST_WARNING_OBJECT (sink,
"correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
sink->priv->avg_skew, mdrift2);
cexternal = cexternal > mdrift ? cexternal - mdrift : 0;
sink->priv->avg_skew -= mdrift;
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
last_align = sink->priv->last_align;
/* if we were aligning in the wrong direction or we aligned more than what we
* will correct, resync */
if (last_align < 0 || last_align > driftsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
crate_num, crate_denom);
} else if (sink->priv->avg_skew < -mdrift2) {
/* master is running faster, move external time forwards */
GST_WARNING_OBJECT (sink,
"correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
sink->priv->avg_skew, -mdrift2);
cexternal += mdrift;
sink->priv->avg_skew += mdrift;
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
last_align = sink->priv->last_align;
/* if we were aligning in the wrong direction or we aligned more than what we
* will correct, resync */
if (last_align > 0 || -last_align > driftsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
crate_num, crate_denom);
}
/* convert, ignoring speed */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
*srender_start = render_start;
*srender_stop = render_stop;
}
/* apply the clock offset but do no slaving otherwise */
static void
gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal, crate_num, crate_denom;
/* get calibration parameters to compensate for offsets */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
/* convert, ignoring speed */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
*srender_start = render_start;
*srender_stop = render_stop;
}
/* converts render_start and render_stop to their slaved values */
static void
gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
switch (sink->priv->slave_method) {
case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
gst_base_audio_sink_resample_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
gst_base_audio_sink_skew_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
case GST_BASE_AUDIO_SINK_SLAVE_NONE:
gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
default:
g_warning ("unknown slaving method %d", sink->priv->slave_method);
break;
}
}
/* must be called with LOCK */
static GstFlowReturn
gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
{
GstClock *clock;
GstClockReturn status;
GstClockTime time, render_delay;
GstFlowReturn ret;
GstBaseAudioSink *sink;
GstClockTime itime, etime;
GstClockTime rate_num, rate_denom;
GstClockTimeDiff jitter;
sink = GST_BASE_AUDIO_SINK (bsink);
clock = GST_ELEMENT_CLOCK (sink);
if (G_UNLIKELY (clock == NULL))
goto no_clock;
/* we provided the global clock, don't need to do anything special */
if (clock == sink->provided_clock)
goto no_slaving;
GST_OBJECT_UNLOCK (sink);
do {
GST_DEBUG_OBJECT (sink, "checking preroll");
ret = gst_base_sink_do_preroll (bsink, obj);
if (ret != GST_FLOW_OK)
goto flushing;
GST_OBJECT_LOCK (sink);
time = sink->priv->us_latency;
GST_OBJECT_UNLOCK (sink);
/* Renderdelay is added onto our own latency, and needs
* to be subtracted as well */
render_delay = gst_base_sink_get_render_delay (bsink);
if (G_LIKELY (time > render_delay))
time -= render_delay;
else
time = 0;
/* preroll done, we can sync since we are in PLAYING now. */
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT, GST_TIME_ARGS (time));
/* wait for the clock, this can be interrupted because we got shut down or
* we PAUSED. */
status = gst_base_sink_wait_clock (bsink, time, &jitter);
GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
GST_TIME_ARGS (jitter));
/* invalid time, no clock or sync disabled, just continue then */
if (status == GST_CLOCK_BADTIME)
break;
/* waiting could have been interrupted and we can be flushing now */
if (G_UNLIKELY (bsink->flushing))
goto flushing;
/* retry if we got unscheduled, which means we did not reach the timeout
* yet. if some other error occures, we continue. */
} while (status == GST_CLOCK_UNSCHEDULED);
GST_OBJECT_LOCK (sink);
GST_DEBUG_OBJECT (sink, "latency synced");
/* when we prerolled in time, we can accurately set the calibration,
* our internal clock should exactly have been the latency (== the running
* time of the external clock) */
etime = GST_ELEMENT_CAST (sink)->base_time + time;
itime = gst_audio_clock_get_time (sink->provided_clock);
itime = gst_audio_clock_adjust (sink->provided_clock, itime);
if (status == GST_CLOCK_EARLY) {
/* when we prerolled late, we have to take into account the lateness */
GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
etime += jitter;
}
/* start ringbuffer so we can start slaving right away when we need to */
gst_ring_buffer_start (sink->ringbuffer);
GST_DEBUG_OBJECT (sink,
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
/* copy the original calibrated rate but update the internal and external
* times. */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
&rate_denom);
gst_clock_set_calibration (sink->provided_clock, itime, etime,
rate_num, rate_denom);
switch (sink->priv->slave_method) {
case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
/* only set as master when we are resampling */
GST_DEBUG_OBJECT (sink, "Setting clock as master");
gst_clock_set_master (sink->provided_clock, clock);
break;
case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
case GST_BASE_AUDIO_SINK_SLAVE_NONE:
default:
break;
}
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
return GST_FLOW_OK;
/* ERRORS */
no_clock:
{
GST_DEBUG_OBJECT (sink, "we have no clock");
return GST_FLOW_OK;
}
no_slaving:
{
GST_DEBUG_OBJECT (sink, "we are not slaved");
return GST_FLOW_OK;
}
flushing:
{
GST_DEBUG_OBJECT (sink, "we are flushing");
GST_OBJECT_LOCK (sink);
return GST_FLOW_WRONG_STATE;
}
}
static gint64
gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink,
GstClockTime sample_offset)
{
GstRingBuffer *ringbuf = sink->ringbuffer;
gint64 align;
gint64 diff;
gint64 maxdrift;
gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
gint64 samples_done = segdone * ringbuf->samples_per_seg;
gint64 headroom = sample_offset - samples_done;
gboolean allow_align = TRUE;
gint rate;
/* now try to align the sample to the previous one, first see how big the
* difference is. */
if (sample_offset >= sink->next_sample)
diff = sample_offset - sink->next_sample;
else
diff = sink->next_sample - sample_offset;
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
/* calculate the max allowed drift in units of samples. By default this is
* 20ms and should be anough to compensate for timestamp rounding errors. */
maxdrift = (rate * sink->priv->drift_tolerance) / GST_MSECOND;
/* calc align with previous sample */
align = sink->next_sample - sample_offset;
/* don't align if it means writing behind the read-segment */
if (diff > headroom && align < 0)
allow_align = FALSE;
if (G_LIKELY (diff < maxdrift && allow_align)) {
GST_DEBUG_OBJECT (sink,
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
G_GINT64_FORMAT, align, maxdrift);
} else {
gint64 diff_s G_GNUC_UNUSED;
/* calculate sample diff in seconds for error message */
diff_s = gst_util_uint64_scale_int (diff, GST_SECOND, rate);
/* timestamps drifted apart from previous samples too much, we need to
* resync. We log this as an element warning. */
GST_WARNING_OBJECT (sink,
"Unexpected discontinuity in audio timestamps of "
"%s%" GST_TIME_FORMAT ", resyncing",
sample_offset > sink->next_sample ? "+" : "-", GST_TIME_ARGS (diff_s));
align = 0;
}
return align;
}
static GstFlowReturn
gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstClockTimeDiff sync_offset, ts_offset;
GstBaseAudioSinkClass *bclass;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
gint64 diff, align;
guint64 ctime, cstop;
gsize offset;
guint8 *data;
gsize size;
guint samples, written;
gint bpf, rate;
gint accum;
gint out_samples;
GstClockTime base_time, render_delay, latency;
GstClock *clock;
gboolean sync, slaved, align_next;
GstFlowReturn ret;
GstSegment clip_seg;
gint64 time_offset;
GstBuffer *out = NULL;
sink = GST_BASE_AUDIO_SINK (bsink);
bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
ringbuf = sink->ringbuffer;
/* can't do anything when we don't have the device */
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf)))
goto wrong_state;
/* Wait for upstream latency before starting the ringbuffer, we do this so
* that we can align the first sample of the ringbuffer to the base_time +
* latency. */
GST_OBJECT_LOCK (sink);
base_time = GST_ELEMENT_CAST (sink)->base_time;
if (G_UNLIKELY (sink->priv->sync_latency)) {
ret = gst_base_audio_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
GST_OBJECT_UNLOCK (sink);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto sync_latency_failed;
/* only do this once until we are set back to PLAYING */
sink->priv->sync_latency = FALSE;
} else {
GST_OBJECT_UNLOCK (sink);
}
/* Before we go on, let's see if we need to payload the data. If yes, we also
* need to unref the output buffer before leaving. */
if (bclass->payload) {
out = bclass->payload (sink, buf);
if (!out)
goto payload_failed;
buf = out;
}
bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
size = gst_buffer_get_size (buf);
if (G_UNLIKELY (size % bpf) != 0)
goto wrong_size;
samples = size / bpf;
out_samples = samples;
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %"
GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset,
GST_TIME_ARGS (bsink->segment.start), samples);
offset = 0;
/* if not valid timestamp or we can't clip or sync, try to play
* sample ASAP */
if (!GST_CLOCK_TIME_IS_VALID (time)) {
render_start = gst_base_audio_sink_get_offset (sink);
render_stop = render_start + samples;
GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
" Using render_start=%" G_GUINT64_FORMAT, size, render_start);
/* we don't have a start so we don't know stop either */
stop = -1;
goto no_sync;
}
/* let's calc stop based on the number of samples in the buffer instead
* of trusting the DURATION */
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rate);
/* prepare the clipping segment. Since we will be subtracting ts-offset and
* device-delay later we scale the start and stop with those values so that we
* can correctly clip them */
clip_seg.format = GST_FORMAT_TIME;
clip_seg.start = bsink->segment.start;
clip_seg.stop = bsink->segment.stop;
clip_seg.duration = -1;
/* the sync offset is the combination of ts-offset and device-delay */
latency = gst_base_sink_get_latency (bsink);
ts_offset = gst_base_sink_get_ts_offset (bsink);
render_delay = gst_base_sink_get_render_delay (bsink);
sync_offset = ts_offset - render_delay + latency;
GST_DEBUG_OBJECT (sink,
"sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
", ts-offset %" G_GINT64_FORMAT, sync_offset,
GST_TIME_ARGS (render_delay), ts_offset);
/* compensate for ts-offset and device-delay when negative we need to
* clip. */
if (sync_offset < 0) {
clip_seg.start += -sync_offset;
if (clip_seg.stop != -1)
clip_seg.stop += -sync_offset;
}
/* samples should be rendered based on their timestamp. All samples
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* boundaries */
if (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop, &ctime,
&cstop))
goto out_of_segment;
/* see if some clipping happened */
diff = ctime - time;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
samples -= diff;
offset += diff * bpf;
time = ctime;
}
diff = stop - cstop;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
samples -= diff;
stop = cstop;
}
/* figure out how to sync */
if ((clock = GST_ELEMENT_CLOCK (bsink)))
sync = bsink->sync;
else
sync = FALSE;
if (!sync) {
/* no sync needed, play sample ASAP */
render_start = gst_base_audio_sink_get_offset (sink);
render_stop = render_start + samples;
GST_DEBUG_OBJECT (sink,
"no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
goto no_sync;
}
/* bring buffer start and stop times to running time */
render_start =
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
render_stop =
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
GST_DEBUG_OBJECT (sink,
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* store the time of the last sample, we'll use this to perform sync on the
* last sample when draining the buffer */
if (bsink->segment.rate >= 0.0) {
sink->priv->eos_time = render_stop;
} else {
sink->priv->eos_time = render_start;
}
/* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */
GST_DEBUG_OBJECT (sink,
"compensating for sync-offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (sync_offset));
render_start += sync_offset;
render_stop += sync_offset;
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
GST_DEBUG_OBJECT (sink,
"after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
if ((slaved = clock != sink->provided_clock)) {
/* handle clock slaving */
gst_base_audio_sink_handle_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
} else {
/* no slaving needed but we need to adapt to the clock calibration
* parameters */
gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
}
GST_DEBUG_OBJECT (sink,
"final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* bring to position in the ringbuffer */
time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
GST_DEBUG_OBJECT (sink,
"time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
if (render_start > time_offset)
render_start -= time_offset;
else
render_start = 0;
if (render_stop > time_offset)
render_stop -= time_offset;
else
render_stop = 0;
/* in some clock slaving cases, all late samples end up at 0 first,
* and subsequent ones align with that until threshold exceeded,
* and then sync back to 0 and so on, so avoid that altogether */
if (G_UNLIKELY (render_start == 0 && render_stop == 0))
goto too_late;
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start, rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop, rate, GST_SECOND);
/* positive playback rate, first sample is render_start, negative rate, first
* sample is render_stop. When no rate conversion is active, render exactly
* the amount of input samples to avoid aligning to rounding errors. */
if (bsink->segment.rate >= 0.0) {
sample_offset = render_start;
if (bsink->segment.rate == 1.0)
render_stop = sample_offset + samples;
} else {
sample_offset = render_stop;
if (bsink->segment.rate == -1.0)
render_start = sample_offset + samples;
}
/* always resync after a discont */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (sink, "resync after discont");
goto no_align;
}
/* resync when we don't know what to align the sample with */
if (G_UNLIKELY (sink->next_sample == -1)) {
GST_DEBUG_OBJECT (sink,
"no align possible: no previous sample position known");
goto no_align;
}
align = gst_base_audio_sink_get_alignment (sink, sample_offset);
sink->priv->last_align = align;
/* apply alignment */
render_start += align;
/* only align stop if we are not slaved to resample */
if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) {
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
goto no_align;
}
render_stop += align;
no_align:
/* number of target samples is difference between start and stop */
out_samples = render_stop - render_start;
no_sync:
/* we render the first or last sample first, depending on the rate */
if (bsink->segment.rate >= 0.0)
sample_offset = render_start;
else
sample_offset = render_stop;
GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
sample_offset, samples, out_samples);
/* we need to accumulate over different runs for when we get interrupted */
accum = 0;
align_next = TRUE;
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
do {
written =
gst_ring_buffer_commit_full (ringbuf, &sample_offset, data + offset,
samples, out_samples, &accum);
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (written == samples)
break;
/* else something interrupted us and we wait for preroll. */
if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
goto stopping;
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
/* update the output samples. FIXME, this will just skip them when pausing
* during trick mode */
if (out_samples > written) {
out_samples -= written;
accum = 0;
} else
break;
samples -= written;
offset += written * bpf;
} while (TRUE);
gst_buffer_unmap (buf, data, size);
if (align_next)
sink->next_sample = sample_offset;
else
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
sink->next_sample);
if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
GST_DEBUG_OBJECT (sink,
"start playback because we are at the end of segment");
gst_ring_buffer_start (ringbuf);
}
ret = GST_FLOW_OK;
done:
if (out)
gst_buffer_unref (out);
return ret;
/* SPECIAL cases */
out_of_segment:
{
GST_DEBUG_OBJECT (sink,
"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
GST_TIME_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (bsink->segment.start));
ret = GST_FLOW_OK;
goto done;
}
too_late:
{
GST_DEBUG_OBJECT (sink, "dropping late sample");
return GST_FLOW_OK;
}
/* ERRORS */
payload_failed:
{
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("failed to payload."));
ret = GST_FLOW_ERROR;
goto done;
}
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
wrong_size:
{
GST_DEBUG_OBJECT (sink, "wrong size");
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
(NULL), ("sink received buffer of wrong size."));
ret = GST_FLOW_ERROR;
goto done;
}
stopping:
{
GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
gst_flow_get_name (ret));
gst_buffer_unmap (buf, data, size);
goto done;
}
sync_latency_failed:
{
GST_DEBUG_OBJECT (sink, "failed waiting for latency");
goto done;
}
}
/**
* gst_base_audio_sink_create_ringbuffer:
* @sink: a #GstBaseAudioSink.
*
* Create and return the #GstRingBuffer for @sink. This function will call the
* ::create_ringbuffer vmethod and will set @sink as the parent of the returned
* buffer (see gst_object_set_parent()).
*
* Returns: The new ringbuffer of @sink.
*/
GstRingBuffer *
gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstBaseAudioSinkClass *bclass;
GstRingBuffer *buffer = NULL;
bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (sink);
if (buffer)
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
return buffer;
}
static void
gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
gpointer user_data)
{
GstBaseSink *basesink;
GstBaseAudioSink *sink;
GstBuffer *buf;
GstFlowReturn ret;
gsize size;
basesink = GST_BASE_SINK (user_data);
sink = GST_BASE_AUDIO_SINK (user_data);
GST_PAD_STREAM_LOCK (basesink->sinkpad);
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
will copy twice, once into data, once into DMA */
GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
" to fill audio buffer", len, basesink->offset);
ret =
gst_pad_pull_range (basesink->sinkpad, basesink->segment.position, len,
&buf);
if (ret != GST_FLOW_OK) {
if (ret == GST_FLOW_EOS)
goto eos;
else
goto error;
}
GST_BASE_SINK_PREROLL_LOCK (basesink);
if (basesink->flushing)
goto flushing;
/* complete preroll and wait for PLAYING */
ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
if (ret != GST_FLOW_OK)
goto preroll_error;
size = gst_buffer_get_size (buf);
if (len != size) {
GST_INFO_OBJECT (basesink,
"got different size than requested from sink pad: %u"
" != %" G_GSIZE_FORMAT, len, size);
len = MIN (size, len);
}
basesink->segment.position += len;
gst_buffer_extract (buf, 0, data, len);
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
error:
{
GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
gst_flow_get_name (ret), ret);
gst_ring_buffer_pause (rbuf);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
eos:
{
/* FIXME: this is not quite correct; we'll be called endlessly until
* the sink gets shut down; maybe we should set a flag somewhere, or
* set segment.stop and segment.duration to the last sample or so */
GST_DEBUG_OBJECT (sink, "EOS");
gst_base_audio_sink_drain (sink);
gst_ring_buffer_pause (rbuf);
gst_element_post_message (GST_ELEMENT_CAST (sink),
gst_message_new_eos (GST_OBJECT_CAST (sink)));
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
}
flushing:
{
GST_DEBUG_OBJECT (sink, "we are flushing");
gst_ring_buffer_pause (rbuf);
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
preroll_error:
{
GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
gst_ring_buffer_pause (rbuf);
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
}
static gboolean
gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
{
gboolean ret;
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
if (active) {
GST_DEBUG_OBJECT (basesink, "activating pull");
gst_ring_buffer_set_callback (sink->ringbuffer,
gst_base_audio_sink_callback, sink);
ret = gst_ring_buffer_activate (sink->ringbuffer, TRUE);
} else {
GST_DEBUG_OBJECT (basesink, "deactivating pull");
gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
ret = gst_ring_buffer_activate (sink->ringbuffer, FALSE);
}
return ret;
}
#if 0
/* should be called with the LOCK */
static GstStateChangeReturn
gst_base_audio_sink_async_play (GstBaseSink * basesink)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (basesink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
if (basesink->pad_mode == GST_PAD_ACTIVATE_PULL) {
/* we always start the ringbuffer in pull mode immediatly */
gst_ring_buffer_start (sink->ringbuffer);
}
return GST_STATE_CHANGE_SUCCESS;
}
#endif
static GstStateChangeReturn
gst_base_audio_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
sink->next_sample = -1;
sink->priv->last_align = -1;
sink->priv->eos_time = -1;
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_ring_buffer_may_start (sink->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 (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
sink->provided_clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
gboolean eos;
GST_OBJECT_LOCK (sink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
eos = GST_BASE_SINK (sink)->eos;
GST_OBJECT_UNLOCK (sink);
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_PAD_ACTIVATE_PULL ||
g_atomic_int_get (&sink->eos_rendering) || eos) {
/* we always start the ringbuffer in pull mode immediatly */
/* sync rendering on eos needs running clock,
* and others need running clock when finished rendering eos */
gst_ring_buffer_start (sink->ringbuffer);
}
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* ringbuffer cannot start anymore */
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_ring_buffer_pause (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
sink->priv->sync_latency = FALSE;
GST_OBJECT_UNLOCK (sink);
break;
case GST_STATE_CHANGE_PAUSED_TO_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 (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
sink->provided_clock));
/* make sure we unblock before calling the parent state change
* so it can grab the STREAM_LOCK */
gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop slaving ourselves to the master, if any */
gst_clock_set_master (sink->provided_clock, NULL);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_ring_buffer_release (sink->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
/* we release again here because the aqcuire happens when setting the
* caps, which happens before we commit the state to PAUSED and thus the
* PAUSED->READY state change (see above, where we release the ringbuffer)
* might not be called when we get here. */
gst_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_ring_buffer_release (sink->ringbuffer);
gst_ring_buffer_close_device (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
GST_OBJECT_UNLOCK (sink);
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (sink, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
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