gstreamer/gst-libs/gst/audio/gstbaseaudiosink.c
Wim Taymans 0990cbf274 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 17:30:17 +00:00

946 lines
28 KiB
C

/* 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
/* 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
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
};
#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 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 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;
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_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_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));
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);
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->async_play =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
}
static void
gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
GstBaseAudioSinkClass * g_class)
{
baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
baseaudiosink->provided_clock = gst_audio_clock_new ("clock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
}
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 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 %llu, delay %u, real %llu, time %"
GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));
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_OBJECT_LOCK (sink);
sink->provide_clock = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (sink);
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:
GST_OBJECT_LOCK (sink);
g_value_set_boolean (value, sink->provide_clock);
GST_OBJECT_UNLOCK (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;
if (!sink->ringbuffer)
return FALSE;
spec = &sink->ringbuffer->spec;
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
/* release old ringbuffer */
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 new ringbuffer");
if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
goto acquire_error;
/* 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_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;
}
/* FIXME, this waits for the drain to happen but it cannot 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 aqcuired the
* ringbuffer. */
if (gst_ring_buffer_is_acquired (sink->ringbuffer))
gst_ring_buffer_start (sink->ringbuffer);
if (sink->next_sample != -1) {
GstClockTime time;
GstClock *clock;
time =
gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
sink->ringbuffer->spec.rate);
GST_OBJECT_LOCK (sink);
if ((clock = GST_ELEMENT_CLOCK (sink)) != NULL) {
GstClockID id = gst_clock_new_single_shot_id (clock, time);
GST_OBJECT_UNLOCK (sink);
GST_DEBUG_OBJECT (sink, "waiting for last sample to play");
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
sink->next_sample = -1;
} else {
GST_OBJECT_UNLOCK (sink);
}
}
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->next_sample = -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 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 GstFlowReturn
gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
guint64 in_offset, clock_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
gint64 diff, align, ctime, cstop;
guint8 *data;
guint size;
guint samples, written;
gint bps;
gint accum;
GstClockTime crate_num;
GstClockTime crate_denom;
gint out_samples;
GstClockTime cinternal, cexternal;
GstClock *clock;
gboolean sync;
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;
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);
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
ringbuf->spec.rate);
GST_DEBUG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", offset %llu, 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);
goto no_sync;
}
/* 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 */
/* let's calc stop based on the number of samples in the buffer instead
* of trusting the DURATION */
if (!gst_segment_clip (&bsink->segment, 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));
/* 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);
clock_offset =
(gst_element_get_base_time (GST_ELEMENT_CAST (bsink)) - cexternal) +
cinternal;
GST_DEBUG_OBJECT (sink, "clock offset %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT
"/%" G_GUINT64_FORMAT, GST_TIME_ARGS (clock_offset), crate_num,
crate_denom);
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start + clock_offset,
ringbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop + clock_offset,
ringbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (sink,
"render: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* 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;
}
if (G_UNLIKELY (sink->next_sample == -1)) {
GST_DEBUG_OBJECT (sink,
"no align possible: no previous sample position known");
goto no_align;
}
if (bsink->segment.rate >= 1.0)
sample_offset = render_start;
else
sample_offset = render_stop;
/* now try to align the sample to the previous one */
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)) {
GST_DEBUG_OBJECT (sink,
"align with prev sample, %" G_GINT64_FORMAT " < %d", diff,
ringbuf->spec.rate / DIFF_TOLERANCE);
/* calc align with previous sample */
align = sink->next_sample - sample_offset;
} 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;
}
/* apply alignment */
render_start += align;
/* only align stop if we are not slaved */
if (clock != sink->provided_clock) {
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:
GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
sink->next_sample, samples, out_samples);
/* we render the first or last sample first, depending on the rate */
if (bsink->segment.rate >= 1.0)
sample_offset = render_start;
else
sample_offset = render_stop;
/* we need to accumulate over different runs for when we get interrupted */
accum = 0;
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 (gst_base_sink_wait_preroll (bsink) != GST_FLOW_OK)
goto stopping;
samples -= written;
data += written * bps;
} while (TRUE);
sink->next_sample = sample_offset;
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, "ringbuffer is stopping");
return GST_FLOW_WRONG_STATE;
}
}
/**
* 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)
{
/* GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (data); */
}
/* should be called with the LOCK */
static GstStateChangeReturn
gst_base_audio_sink_async_play (GstBaseSink * basesink)
{
GstClock *clock;
GstClockTime time, base;
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (basesink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
clock = GST_ELEMENT_CLOCK (sink);
if (clock == NULL)
goto no_clock;
/* FIXME, only start slaving when we really start the ringbuffer */
/* if we are slaved to a clock, we need to set the initial
* calibration */
if (clock != sink->provided_clock) {
GstClockTime rate_num, rate_denom;
base = GST_ELEMENT_CAST (sink)->base_time;
time = gst_clock_get_internal_time (sink->provided_clock);
GST_DEBUG_OBJECT (sink,
"time: %" GST_TIME_FORMAT " base: %" GST_TIME_FORMAT,
GST_TIME_ARGS (time), GST_TIME_ARGS (base));
/* FIXME, this is not yet accurate enough for smooth playback */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
&rate_denom);
/* Does not work yet. */
gst_clock_set_calibration (sink->provided_clock, time, base,
rate_num, rate_denom);
gst_clock_set_master (sink->provided_clock, clock);
}
no_clock:
return GST_STATE_CHANGE_SUCCESS;
}
static GstStateChangeReturn
gst_base_audio_sink_do_play (GstBaseAudioSink * sink)
{
GstStateChangeReturn ret;
GST_OBJECT_LOCK (sink);
ret = gst_base_audio_sink_async_play (GST_BASE_SINK_CAST (sink));
GST_OBJECT_UNLOCK (sink);
return ret;
}
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) {
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
gst_ring_buffer_set_callback (sink->ringbuffer,
gst_base_audio_sink_callback, sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
sink->next_sample = -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_base_audio_sink_do_play (sink);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* need to take the lock so we don't interfere with an
* async play */
GST_OBJECT_LOCK (sink);
/* ringbuffer cannot start anymore */
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_ring_buffer_pause (sink->ringbuffer);
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:
/* slop 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_release (sink->ringbuffer);
gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_ring_buffer_close_device (sink->ringbuffer);
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;
}
}