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gstreamer/gst-libs/gst/audio/gstbaseaudiosink.c
Wim Taymans 57a13f28de audiosink: free the ringbuffer when going to NULL 
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
2009-06-17 13:18:18 +02:00

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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;
};
/* BaseAudioSink signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
/* we tollerate half a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position.
* This is an emergency resync fallback since buffers marked as DISCONT will
* always lock to the correct timestamp immediatly and buffers not marked as
* DISCONT are contiguous by definition.
*/
#define DIFF_TOLERANCE 2
/* 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
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
PROP_SLAVE_METHOD,
PROP_CAN_ACTIVATE_PULL
};
GType
gst_base_audio_sink_slave_method_get_type (void)
{
static GType slave_method_type = 0;
static const GEnumValue slave_method[] = {
{GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
{GST_BASE_AUDIO_SINK_SLAVE_SKEW, "Skew slaving", "skew"},
{GST_BASE_AUDIO_SINK_SLAVE_NONE, "No slaving", "none"},
{0, NULL, NULL},
};
if (!slave_method_type) {
slave_method_type =
g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
}
return slave_method_type;
}
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_base_audio_sink, GstBaseSink,
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);
static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
basesink);
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 (GstPad * pad, GstQuery * query);
/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
static void
gst_base_audio_sink_base_init (gpointer g_class)
{
}
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_DEBUG_FUNCPTR (gst_base_audio_sink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property);
gobject_class->dispose = GST_DEBUG_FUNCPTR (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));
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->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);
gstbasesink_class->async_play =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
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,
GstBaseAudioSinkClass * g_class)
{
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->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
GST_BASE_SINK (baseaudiosink)->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
/* install some custom pad_query functions */
gst_pad_set_query_function (GST_BASE_SINK_PAD (baseaudiosink),
GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad));
}
static void
gst_base_audio_sink_dispose (GObject * object)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (object);
if (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 (GstPad * pad, GstQuery * query)
{
gboolean res = FALSE;
GstBaseAudioSink *basesink;
basesink = GST_BASE_AUDIO_SINK (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_LOG_OBJECT (pad, "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:
break;
}
gst_object_unref (basesink);
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");
if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) {
GST_DEBUG_OBJECT (basesink,
"we are not yet negotiated, can't report latency yet");
res = FALSE;
goto done;
}
/* 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 min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
GstRingBufferSpec *spec;
spec = &basesink->ringbuffer->spec;
basesink->priv->us_latency = min_l;
min_latency =
gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
GST_SECOND, spec->rate * spec->bytes_per_sample);
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = min_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency = min_latency + (max_l == -1 ? 0 : 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));
} 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.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.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;
}
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;
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;
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;
if (!sink->ringbuffer)
return FALSE;
spec = &sink->ringbuffer->spec;
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
/* 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_ACTIVATE_PUSH) {
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
gst_ring_buffer_activate (sink->ringbuffer, TRUE);
}
/* 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->rate * spec->bytes_per_sample);
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.rate)
return TRUE;
/* 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");
}
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;
case GST_EVENT_NEWSEGMENT:
{
gdouble rate;
/* we only need the rate */
gst_event_parse_new_segment_full (event, NULL, &rate, NULL, NULL,
NULL, NULL, NULL);
GST_DEBUG_OBJECT (sink, "new segment rate of %f", rate);
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, segtime, segtime2;
gint segsamples;
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);
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 is the length of a segment */
segtime = sink->ringbuffer->spec.latency_time * 1000;
segtime2 = segtime / 2;
/* adjust playout pointer based on skew */
if (sink->priv->avg_skew > segtime2) {
/* 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, segtime2);
cexternal = cexternal > segtime ? cexternal - segtime : 0;
sink->priv->avg_skew -= segtime;
segsamples =
sink->ringbuffer->spec.segsize /
sink->ringbuffer->spec.bytes_per_sample;
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 > segsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " segsamples %u, next %"
G_GUINT64_FORMAT, last_align, segsamples, sink->next_sample);
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
crate_num, crate_denom);
} else if (sink->priv->avg_skew < -segtime2) {
/* 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, -segtime2);
cexternal += segtime;
sink->priv->avg_skew += segtime;
segsamples =
sink->ringbuffer->spec.segsize /
sink->ringbuffer->spec.bytes_per_sample;
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 > segsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " segsamples %u, next %"
G_GUINT64_FORMAT, last_align, segsamples, 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;
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);
/* 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);
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 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;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
gint64 diff, align, ctime, cstop;
guint8 *data;
guint size;
guint samples, written;
gint bps;
gint accum;
gint out_samples;
GstClockTime base_time, render_delay, latency;
GstClock *clock;
gboolean sync, slaved, align_next;
GstFlowReturn ret;
GstSegment clip_seg;
sink = GST_BASE_AUDIO_SINK (bsink);
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);
}
bps = ringbuf->spec.bytes_per_sample;
size = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (size % bps) != 0)
goto wrong_size;
samples = size / bps;
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);
data = GST_BUFFER_DATA (buf);
/* 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 %u has no time. Using render_start=%" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (buf), 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,
ringbuf->spec.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, ringbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
samples -= diff;
data += diff * bps;
time = ctime;
}
diff = stop - cstop;
if (diff > 0) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, ringbuf->spec.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);
}
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start,
ringbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop,
ringbuf->spec.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;
}
/* 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;
/* we tollerate half a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position. We always resync if we got a discont anyway and
* non-discont should be aligned by definition. */
if (G_LIKELY (diff < ringbuf->spec.rate / DIFF_TOLERANCE)) {
/* calc align with previous sample */
align = sink->next_sample - sample_offset;
GST_DEBUG_OBJECT (sink,
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align,
ringbuf->spec.rate / DIFF_TOLERANCE);
} else {
/* bring sample diff to seconds for error message */
diff = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate);
/* timestamps drifted apart from previous samples too much, we need to
* resync. We log this as an element warning. */
GST_ELEMENT_WARNING (sink, CORE, CLOCK,
("Compensating for audio synchronisation problems"),
("Unexpected discontinuity in audio timestamps of more "
"than half a second (%" GST_TIME_FORMAT "), resyncing",
GST_TIME_ARGS (diff)));
align = 0;
}
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;
do {
written =
gst_ring_buffer_commit_full (ringbuf, &sample_offset, data, 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;
data += written * bps;
} while (TRUE);
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);
}
return GST_FLOW_OK;
/* 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));
return GST_FLOW_OK;
}
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
wrong_size:
{
GST_DEBUG_OBJECT (sink, "wrong size");
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
(NULL), ("sink received buffer of wrong size."));
return GST_FLOW_ERROR;
}
stopping:
{
GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
gst_flow_get_name (ret));
return ret;
}
sync_latency_failed:
{
GST_DEBUG_OBJECT (sink, "failed waiting for latency");
return ret;
}
}
/**
* 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;
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 %d bytes offset %" G_GUINT64_FORMAT
" to fill audio buffer", len, basesink->offset);
ret =
gst_pad_pull_range (basesink->sinkpad, basesink->segment.last_stop, len,
&buf);
if (ret != GST_FLOW_OK) {
if (ret == GST_FLOW_UNEXPECTED)
goto eos;
else
goto error;
}
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
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;
if (len != GST_BUFFER_SIZE (buf)) {
GST_INFO_OBJECT (basesink,
"got different size than requested from sink pad: %u != %u", len,
GST_BUFFER_SIZE (buf));
len = MIN (GST_BUFFER_SIZE (buf), len);
}
basesink->segment.last_stop += len;
memcpy (data, GST_BUFFER_DATA (buf), len);
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
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_PAD_PREROLL_UNLOCK (basesink->sinkpad);
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_PAD_PREROLL_UNLOCK (basesink->sinkpad);
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;
}
/* 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_ACTIVATE_PULL) {
/* we always start the ringbuffer in pull mode immediatly */
gst_ring_buffer_start (sink->ringbuffer);
}
return GST_STATE_CHANGE_SUCCESS;
}
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);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_OBJECT_LOCK (sink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
GST_OBJECT_UNLOCK (sink);
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL) {
/* we always start the ringbuffer in pull mode immediatly */
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:
/* 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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