gstreamer/gst-libs/gst/audio/gstbaseaudiosink.c
Wim Taymans a878cbdfe1 gst-libs/gst/audio/gstbaseaudiosink.c: Remove g_print
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Remove g_print
Use sync property from baseclass to disable sync.
2005-10-24 14:59:55 +00:00

564 lines
16 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.
*/
#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 a 10th of a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position. */
#define DIFF_TOLERANCE 10
#define DEFAULT_BUFFER_TIME 500 * GST_USECOND
#define DEFAULT_LATENCY_TIME 10 * GST_USECOND
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
};
#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_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 (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in milliseconds (-1 = default)",
-1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"Audio latency in milliseconds (-1 = default)",
-1, G_MAXINT64, DEFAULT_LATENCY_TIME, 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);
}
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->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->clock)
gst_object_unref (sink->clock);
sink->clock = NULL;
if (sink->ringbuffer)
gst_object_unref (sink->ringbuffer);
sink->ringbuffer = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstClock *
gst_base_audio_sink_provide_clock (GstElement * elem)
{
GstBaseAudioSink *sink;
sink = GST_BASE_AUDIO_SINK (elem);
return GST_CLOCK (gst_object_ref (sink->clock));
}
static GstClockTime
gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
{
guint64 samples;
GstClockTime result;
if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
return GST_CLOCK_TIME_NONE;
/* our processed samples are always increasing */
samples = gst_ring_buffer_samples_done (sink->ringbuffer);
result = samples * GST_SECOND / sink->ringbuffer->spec.rate;
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;
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;
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;
spec = &sink->ringbuffer->spec;
GST_DEBUG ("release old ringbuffer");
/* release old ringbuffer */
gst_ring_buffer_release (sink->ringbuffer);
GST_DEBUG ("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 ("acquire new ringbuffer");
if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
goto acquire_error;
/* calculate actual latency and buffer times */
spec->latency_time =
spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample);
spec->buffer_time =
spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
spec->bytes_per_sample);
gst_ring_buffer_debug_spec_buff (spec);
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
acquire_error:
{
GST_DEBUG ("could not acquire ringbuffer");
return FALSE;
}
}
static void
gst_base_audio_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;
}
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:
gst_ring_buffer_pause (sink->ringbuffer);
gst_ring_buffer_clear_all (sink->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
sink->next_sample = -1;
gst_ring_buffer_clear_all (sink->ringbuffer);
break;
case GST_EVENT_EOS:
gst_ring_buffer_start (sink->ringbuffer);
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 ("ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), (NULL));
return GST_FLOW_ERROR;
}
}
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;
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 render_offset, in_offset;
GstClockTime time, render_time, duration;
GstClockTimeDiff render_diff;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
gint64 diff;
guint8 *data;
guint size;
guint samples;
gint bps;
sink = GST_BASE_AUDIO_SINK (bsink);
ringbuf = sink->ringbuffer;
/* can't do anything when we don't have the device */
if (!gst_ring_buffer_is_acquired (ringbuf))
goto wrong_state;
bps = ringbuf->spec.bytes_per_sample;
size = GST_BUFFER_SIZE (buf);
if (size % bps != 0)
goto wrong_size;
samples = size / bps;
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
duration = GST_BUFFER_DURATION (buf);
data = GST_BUFFER_DATA (buf);
GST_DEBUG ("time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT,
GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment_start));
/* if not valid timestamp or we don't need to sync, try to play
* sample ASAP */
if (!GST_CLOCK_TIME_IS_VALID (time) || !bsink->sync) {
render_offset = gst_base_audio_sink_get_offset (sink);
goto no_sync;
}
render_diff = time - bsink->segment_start;
/* samples should be rendered based on their timestamp. All samples
* arriving before the segment_start are to be thrown away */
/* FIXME, for now we drop the sample completely, we should
* in fact clip the sample. Same for the segment_stop, actually. */
if (render_diff < 0)
goto out_of_segment;
/* bring buffer timestamp to stream time */
render_time = render_diff;
/* adjust for rate */
render_time /= ABS (bsink->segment_rate);
/* adjust for accumulated segments */
render_time += bsink->segment_accum;
/* add base time to get absolute clock time */
render_time += gst_element_get_base_time (GST_ELEMENT (bsink));
/* and bring the time to the offset in the buffer */
render_offset = render_time * ringbuf->spec.rate / GST_SECOND;
/* roundoff errors in timestamp conversion */
if (sink->next_sample != -1)
diff = ABS ((gint64) render_offset - (gint64) sink->next_sample);
else
diff = ringbuf->spec.rate;
GST_DEBUG ("render time %" GST_TIME_FORMAT
", render offset %llu, diff %lld, samples %lu",
GST_TIME_ARGS (render_time), render_offset, diff, samples);
/* we tollerate a 10th of a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position. */
if (diff < ringbuf->spec.rate / DIFF_TOLERANCE) {
GST_DEBUG ("align with prev sample, %" G_GINT64_FORMAT " < %lu", diff,
ringbuf->spec.rate / DIFF_TOLERANCE);
/* just align with previous sample then */
render_offset = sink->next_sample;
} else {
GST_DEBUG ("resync");
}
no_sync:
/* clip length based on rate */
samples = MIN (samples, samples / ABS (bsink->segment_rate));
/* the next sample should be current sample and its length */
sink->next_sample = render_offset + samples;
gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
if (GST_CLOCK_TIME_IS_VALID (time) && time + duration >= bsink->segment_stop) {
GST_DEBUG ("start playback because we are at the end of segment");
gst_ring_buffer_start (ringbuf);
}
return GST_FLOW_OK;
out_of_segment:
{
GST_DEBUG ("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;
}
wrong_state:
{
GST_DEBUG ("ringbuffer not negotiated");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
wrong_size:
{
GST_DEBUG ("wrong size");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink received buffer of wrong size."),
("sink received buffer of wrong size."));
return GST_FLOW_ERROR;
}
}
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);
}
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))
return GST_STATE_CHANGE_FAILURE;
sink->next_sample = 0;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_ring_buffer_pause (sink->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_stop (sink->ringbuffer);
gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
gst_ring_buffer_release (sink->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_ring_buffer_close_device (sink->ringbuffer);
break;
default:
break;
}
return ret;
}