gstreamer/gst/audiobuffer/gstaudioringbuffer.c
2012-04-05 18:02:56 +02:00

1181 lines
34 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>
*
* gstaudioringbuffer.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:element-audioringbuffer
* @short_description: Asynchronous audio ringbuffer.
*
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <glib/gstdio.h>
#include <gst/gst.h>
#include <gst/gst-i18n-plugin.h>
#include <gst/audio/gstringbuffer.h>
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
GST_DEBUG_CATEGORY_STATIC (audioringbuffer_debug);
#define GST_CAT_DEFAULT (audioringbuffer_debug)
enum
{
LAST_SIGNAL
};
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_SEGMENT_TIME ((10 * GST_MSECOND) / GST_USECOND)
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_SEGMENT_TIME,
PROP_LAST
};
#define GST_TYPE_AUDIO_RINGBUFFER \
(gst_audio_ringbuffer_get_type())
#define GST_AUDIO_RINGBUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbuffer))
#define GST_AUDIO_RINGBUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbufferClass))
#define GST_IS_AUDIO_RINGBUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RINGBUFFER))
#define GST_IS_AUDIO_RINGBUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RINGBUFFER))
#define GST_AUDIO_RINGBUFFER_CAST(obj) \
((GstAudioRingbuffer *)(obj))
static GType gst_audio_ringbuffer_get_type (void);
typedef struct _GstAudioRingbuffer GstAudioRingbuffer;
typedef struct _GstAudioRingbufferClass GstAudioRingbufferClass;
typedef struct _GstIntRingBuffer GstIntRingBuffer;
typedef struct _GstIntRingBufferClass GstIntRingBufferClass;
struct _GstAudioRingbuffer
{
GstElement element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
gboolean pushing;
gboolean pulling;
/* segments to keep track of timestamps */
GstSegment sink_segment;
GstSegment src_segment;
/* flowreturn when srcpad is paused */
gboolean is_eos;
gboolean flushing;
gboolean waiting;
GCond *cond;
GstRingBuffer *buffer;
GstClockTime buffer_time;
GstClockTime segment_time;
guint64 next_sample;
guint64 last_align;
};
struct _GstAudioRingbufferClass
{
GstElementClass parent_class;
};
#define GST_TYPE_INT_RING_BUFFER (gst_int_ring_buffer_get_type())
#define GST_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_INT_RING_BUFFER,GstIntRingBuffer))
#define GST_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_INT_RING_BUFFER,GstIntRingBufferClass))
#define GST_INT_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_INT_RING_BUFFER, GstIntRingBufferClass))
#define GST_INT_RING_BUFFER_CAST(obj) ((GstIntRingBuffer *)obj)
#define GST_IS_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_INT_RING_BUFFER))
#define GST_IS_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_INT_RING_BUFFER))
struct _GstIntRingBuffer
{
GstRingBuffer object;
};
struct _GstIntRingBufferClass
{
GstRingBufferClass parent_class;
};
GST_BOILERPLATE (GstIntRingBuffer, gst_int_ring_buffer, GstRingBuffer,
GST_TYPE_RING_BUFFER);
static gboolean
gst_int_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
spec->seglatency = spec->segtotal;
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
return TRUE;
}
static gboolean
gst_int_ring_buffer_release (GstRingBuffer * buf)
{
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_int_ring_buffer_start (GstRingBuffer * buf)
{
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (buf));
GST_OBJECT_LOCK (ringbuffer);
if (G_UNLIKELY (ringbuffer->waiting)) {
ringbuffer->waiting = FALSE;
GST_DEBUG_OBJECT (ringbuffer, "start, sending signal");
g_cond_broadcast (ringbuffer->cond);
}
GST_OBJECT_UNLOCK (ringbuffer);
return TRUE;
}
static void
gst_int_ring_buffer_base_init (gpointer klass)
{
}
static void
gst_int_ring_buffer_class_init (GstIntRingBufferClass * klass)
{
GstRingBufferClass *gstringbuffer_class;
gstringbuffer_class = (GstRingBufferClass *) klass;
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_int_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_int_ring_buffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_int_ring_buffer_start);
}
static void
gst_int_ring_buffer_init (GstIntRingBuffer * buff,
GstIntRingBufferClass * g_class)
{
}
static GstRingBuffer *
gst_int_ring_buffer_new (void)
{
GstRingBuffer *res;
res = g_object_new (GST_TYPE_INT_RING_BUFFER, NULL);
return res;
}
/* can't use boilerplate as we need to register with Queue2 to avoid conflicts
* with ringbuffer in core elements */
static void gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass);
static void gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer,
GstAudioRingbufferClass * g_class);
static GstElementClass *elem_parent_class;
static GType
gst_audio_ringbuffer_get_type (void)
{
static GType gst_audio_ringbuffer_type = 0;
if (!gst_audio_ringbuffer_type) {
static const GTypeInfo gst_audio_ringbuffer_info = {
sizeof (GstAudioRingbufferClass),
NULL,
NULL,
(GClassInitFunc) gst_audio_ringbuffer_class_init,
NULL,
NULL,
sizeof (GstAudioRingbuffer),
0,
(GInstanceInitFunc) gst_audio_ringbuffer_init,
NULL
};
gst_audio_ringbuffer_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstAudioRingbuffer",
&gst_audio_ringbuffer_info, 0);
}
return gst_audio_ringbuffer_type;
}
static void gst_audio_ringbuffer_finalize (GObject * object);
static void gst_audio_ringbuffer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_ringbuffer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstFlowReturn gst_audio_ringbuffer_chain (GstPad * pad,
GstBuffer * buffer);
static GstFlowReturn gst_audio_ringbuffer_bufferalloc (GstPad * pad,
guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
static gboolean gst_audio_ringbuffer_handle_sink_event (GstPad * pad,
GstEvent * event);
static gboolean gst_audio_ringbuffer_handle_src_event (GstPad * pad,
GstEvent * event);
static gboolean gst_audio_ringbuffer_handle_src_query (GstPad * pad,
GstQuery * query);
static GstCaps *gst_audio_ringbuffer_getcaps (GstPad * pad);
static gboolean gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_audio_ringbuffer_get_range (GstPad * pad,
guint64 offset, guint length, GstBuffer ** buffer);
static gboolean gst_audio_ringbuffer_src_checkgetrange_function (GstPad * pad);
static gboolean gst_audio_ringbuffer_src_activate_pull (GstPad * pad,
gboolean active);
static gboolean gst_audio_ringbuffer_src_activate_push (GstPad * pad,
gboolean active);
static gboolean gst_audio_ringbuffer_sink_activate_push (GstPad * pad,
gboolean active);
static GstStateChangeReturn gst_audio_ringbuffer_change_state (GstElement *
element, GstStateChange transition);
/* static guint gst_audio_ringbuffer_signals[LAST_SIGNAL] = { 0 }; */
static void
gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
elem_parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_property);
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in nanoseconds", 1,
G_MAXINT64, DEFAULT_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SEGMENT_TIME,
g_param_spec_int64 ("segment-time", "Segment Time",
"Audio segment duration in nanoseconds", 1,
G_MAXINT64, DEFAULT_SEGMENT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&srctemplate));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sinktemplate));
gst_element_class_set_details_simple (gstelement_class, "AudioRingbuffer",
"Generic",
"Asynchronous Audio ringbuffer", "Wim Taymans <wim.taymans@gmail.com>");
/* set several parent class virtual functions */
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_finalize);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_change_state);
}
static void
gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer,
GstAudioRingbufferClass * g_class)
{
ringbuffer->sinkpad =
gst_pad_new_from_static_template (&sinktemplate, "sink");
gst_pad_set_chain_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_chain));
gst_pad_set_activatepush_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_sink_activate_push));
gst_pad_set_event_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_sink_event));
gst_pad_set_getcaps_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps));
gst_pad_set_setcaps_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_setcaps));
gst_pad_set_bufferalloc_function (ringbuffer->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_bufferalloc));
gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->sinkpad);
ringbuffer->srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
gst_pad_set_activatepull_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_pull));
gst_pad_set_activatepush_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_push));
gst_pad_set_getrange_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_range));
gst_pad_set_checkgetrange_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_checkgetrange_function));
gst_pad_set_getcaps_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps));
gst_pad_set_event_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_event));
gst_pad_set_query_function (ringbuffer->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_query));
gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->srcpad);
gst_segment_init (&ringbuffer->sink_segment, GST_FORMAT_TIME);
ringbuffer->cond = g_cond_new ();
ringbuffer->is_eos = FALSE;
ringbuffer->buffer_time = DEFAULT_BUFFER_TIME;
ringbuffer->segment_time = DEFAULT_SEGMENT_TIME;
GST_DEBUG_OBJECT (ringbuffer,
"initialized ringbuffer's not_empty & not_full conditions");
}
/* called only once, as opposed to dispose */
static void
gst_audio_ringbuffer_finalize (GObject * object)
{
GstAudioRingbuffer *ringbuffer = GST_AUDIO_RINGBUFFER (object);
GST_DEBUG_OBJECT (ringbuffer, "finalizing ringbuffer");
g_cond_free (ringbuffer->cond);
G_OBJECT_CLASS (elem_parent_class)->finalize (object);
}
static GstCaps *
gst_audio_ringbuffer_getcaps (GstPad * pad)
{
GstAudioRingbuffer *ringbuffer;
GstPad *otherpad;
GstCaps *result;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
otherpad =
(pad == ringbuffer->srcpad ? ringbuffer->sinkpad : ringbuffer->srcpad);
result = gst_pad_peer_get_caps (otherpad);
if (result == NULL)
result = gst_caps_new_any ();
return result;
}
static gboolean
gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioRingbuffer *ringbuffer;
GstRingBufferSpec *spec;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
if (!ringbuffer->buffer)
return FALSE;
spec = &ringbuffer->buffer->spec;
GST_DEBUG_OBJECT (ringbuffer, "release old ringbuffer");
/* release old ringbuffer */
gst_ring_buffer_activate (ringbuffer->buffer, FALSE);
gst_ring_buffer_release (ringbuffer->buffer);
GST_DEBUG_OBJECT (ringbuffer, "parse caps");
spec->buffer_time = ringbuffer->buffer_time;
spec->latency_time = ringbuffer->segment_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 (ringbuffer, "acquire ringbuffer");
if (!gst_ring_buffer_acquire (ringbuffer->buffer, spec))
goto acquire_error;
GST_DEBUG_OBJECT (ringbuffer, "activate ringbuffer");
gst_ring_buffer_activate (ringbuffer->buffer, 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 (ringbuffer, "could not parse caps");
GST_ELEMENT_ERROR (ringbuffer, STREAM, FORMAT,
(NULL), ("cannot parse audio format."));
return FALSE;
}
acquire_error:
{
GST_DEBUG_OBJECT (ringbuffer, "could not acquire ringbuffer");
return FALSE;
}
}
static GstFlowReturn
gst_audio_ringbuffer_bufferalloc (GstPad * pad, guint64 offset, guint size,
GstCaps * caps, GstBuffer ** buf)
{
GstAudioRingbuffer *ringbuffer;
GstFlowReturn result;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
/* Forward to src pad, without setting caps on the src pad */
result = gst_pad_alloc_buffer (ringbuffer->srcpad, offset, size, caps, buf);
return result;
}
static gboolean
gst_audio_ringbuffer_handle_sink_event (GstPad * pad, GstEvent * event)
{
GstAudioRingbuffer *ringbuffer;
gboolean forward;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (pad));
forward = ringbuffer->pushing || ringbuffer->pulling;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
{
GST_LOG_OBJECT (ringbuffer, "received flush start event");
break;
}
case GST_EVENT_FLUSH_STOP:
{
ringbuffer->is_eos = FALSE;
GST_LOG_OBJECT (ringbuffer, "received flush stop event");
break;
}
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate, arate;
GstFormat format;
gint64 start, stop, time;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
gst_segment_set_newsegment_full (&ringbuffer->sink_segment, update, rate,
arate, format, start, stop, time);
break;
}
case GST_EVENT_EOS:
ringbuffer->is_eos = TRUE;
break;
default:
break;
}
if (forward) {
gst_pad_push_event (ringbuffer->srcpad, event);
} else {
if (event)
gst_event_unref (event);
}
return TRUE;
}
#define DIFF_TOLERANCE 2
static GstFlowReturn
gst_audio_ringbuffer_render (GstAudioRingbuffer * ringbuffer, GstBuffer * buf)
{
GstRingBuffer *rbuf;
gint bps, accum;
guint size;
guint samples, written, out_samples;
gint64 diff, align, ctime, cstop;
guint8 *data;
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
gboolean align_next;
rbuf = ringbuffer->buffer;
/* can't do anything when we don't have the device */
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (rbuf)))
goto wrong_state;
bps = rbuf->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 (ringbuffer,
"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
", samples %u", GST_TIME_ARGS (time), in_offset,
GST_TIME_ARGS (ringbuffer->sink_segment.start), samples);
data = GST_BUFFER_DATA (buf);
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
rbuf->spec.rate);
if (!gst_segment_clip (&ringbuffer->sink_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, rbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (ringbuffer, "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, rbuf->spec.rate, GST_SECOND);
GST_DEBUG_OBJECT (ringbuffer, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
samples -= diff;
stop = cstop;
}
/* bring buffer start and stop times to running time */
render_start =
gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME,
time);
render_stop =
gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME,
stop);
GST_DEBUG_OBJECT (ringbuffer,
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start,
rbuf->spec.rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop,
rbuf->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 (ringbuffer->sink_segment.rate >= 0.0) {
sample_offset = render_start;
if (ringbuffer->sink_segment.rate == 1.0)
render_stop = sample_offset + samples;
} else {
sample_offset = render_stop;
if (ringbuffer->sink_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 (ringbuffer, "resync after discont");
goto no_align;
}
/* resync when we don't know what to align the sample with */
if (G_UNLIKELY (ringbuffer->next_sample == -1)) {
GST_DEBUG_OBJECT (ringbuffer,
"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 >= ringbuffer->next_sample)
diff = sample_offset - ringbuffer->next_sample;
else
diff = ringbuffer->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 < rbuf->spec.rate / DIFF_TOLERANCE)) {
/* calc align with previous sample */
align = ringbuffer->next_sample - sample_offset;
GST_DEBUG_OBJECT (ringbuffer,
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align,
rbuf->spec.rate / DIFF_TOLERANCE);
} else {
/* bring sample diff to seconds for error message */
diff = gst_util_uint64_scale_int (diff, GST_SECOND, rbuf->spec.rate);
/* timestamps drifted apart from previous samples too much, we need to
* resync. We log this as an element warning. */
GST_ELEMENT_WARNING (ringbuffer, 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;
}
ringbuffer->last_align = align;
/* apply alignment */
render_start += align;
render_stop += align;
no_align:
/* number of target samples is difference between start and stop */
out_samples = render_stop - render_start;
/* we render the first or last sample first, depending on the rate */
if (ringbuffer->sink_segment.rate >= 0.0)
sample_offset = render_start;
else
sample_offset = render_stop;
GST_DEBUG_OBJECT (ringbuffer, "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 (rbuf, &sample_offset, data, samples,
out_samples, &accum);
GST_DEBUG_OBJECT (ringbuffer, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (written == samples)
break;
GST_OBJECT_LOCK (ringbuffer);
if (ringbuffer->flushing)
goto flushing;
GST_OBJECT_UNLOCK (ringbuffer);
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
samples -= written;
data += written * bps;
} while (TRUE);
if (align_next)
ringbuffer->next_sample = sample_offset;
else
ringbuffer->next_sample = -1;
GST_DEBUG_OBJECT (ringbuffer, "next sample expected at %" G_GUINT64_FORMAT,
ringbuffer->next_sample);
if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= ringbuffer->sink_segment.stop) {
GST_DEBUG_OBJECT (ringbuffer,
"start playback because we are at the end of segment");
gst_ring_buffer_start (rbuf);
}
return GST_FLOW_OK;
/* SPECIAL cases */
out_of_segment:
{
GST_DEBUG_OBJECT (ringbuffer,
"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
GST_TIME_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (ringbuffer->sink_segment.start));
return GST_FLOW_OK;
}
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (ringbuffer, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (ringbuffer, STREAM, FORMAT, (NULL),
("ringbuffer not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
wrong_size:
{
GST_DEBUG_OBJECT (ringbuffer, "wrong size");
GST_ELEMENT_ERROR (ringbuffer, STREAM, WRONG_TYPE,
(NULL), ("ringbuffer received buffer of wrong size."));
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (ringbuffer, "ringbuffer is flushing");
GST_OBJECT_UNLOCK (ringbuffer);
return GST_FLOW_FLUSHING;
}
}
static GstFlowReturn
gst_audio_ringbuffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstFlowReturn res;
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (pad));
if (ringbuffer->pushing) {
GST_DEBUG_OBJECT (ringbuffer, "proxy pushing buffer");
res = gst_pad_push (ringbuffer->srcpad, buffer);
} else {
GST_DEBUG_OBJECT (ringbuffer, "render buffer in ringbuffer");
res = gst_audio_ringbuffer_render (ringbuffer, buffer);
}
return res;
}
static gboolean
gst_audio_ringbuffer_handle_src_event (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstAudioRingbuffer *ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
/* just forward upstream */
res = gst_pad_push_event (ringbuffer->sinkpad, event);
return res;
}
static gboolean
gst_audio_ringbuffer_handle_src_query (GstPad * pad, GstQuery * query)
{
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
break;
case GST_QUERY_DURATION:
break;
case GST_QUERY_BUFFERING:
break;
default:
break;
}
return TRUE;
}
static GstFlowReturn
gst_audio_ringbuffer_get_range (GstPad * pad, guint64 offset, guint length,
GstBuffer ** buffer)
{
GstAudioRingbuffer *ringbuffer;
GstRingBuffer *rbuf;
GstFlowReturn ret;
ringbuffer = GST_AUDIO_RINGBUFFER_CAST (gst_pad_get_parent (pad));
rbuf = ringbuffer->buffer;
if (ringbuffer->pulling) {
GST_DEBUG_OBJECT (ringbuffer, "proxy pulling range");
ret = gst_pad_pull_range (ringbuffer->sinkpad, offset, length, buffer);
} else {
guint8 *data;
guint len;
guint64 sample;
gint bps, segsize, segtotal, sps;
gint sampleslen, segdone;
gint readseg, sampleoff;
guint8 *dest;
GST_DEBUG_OBJECT (ringbuffer,
"pulling data at %" G_GUINT64_FORMAT ", length %u", offset, length);
if (offset != ringbuffer->src_segment.last_stop) {
GST_DEBUG_OBJECT (ringbuffer, "expected offset %" G_GINT64_FORMAT,
ringbuffer->src_segment.last_stop);
}
/* first wait till we have something in the ringbuffer and it
* is running */
GST_OBJECT_LOCK (ringbuffer);
if (ringbuffer->flushing)
goto flushing;
while (ringbuffer->waiting) {
GST_DEBUG_OBJECT (ringbuffer, "waiting for unlock");
g_cond_wait (ringbuffer->cond, GST_OBJECT_GET_LOCK (ringbuffer));
GST_DEBUG_OBJECT (ringbuffer, "unlocked");
if (ringbuffer->flushing)
goto flushing;
}
GST_OBJECT_UNLOCK (ringbuffer);
bps = rbuf->spec.bytes_per_sample;
if (G_UNLIKELY (length % bps) != 0)
goto wrong_size;
segsize = rbuf->spec.segsize;
segtotal = rbuf->spec.segtotal;
sps = rbuf->samples_per_seg;
dest = GST_BUFFER_DATA (rbuf->data);
sample = offset / bps;
len = length / bps;
*buffer = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (*buffer);
while (len) {
gint diff;
/* figure out the segment and the offset inside the segment where
* the sample should be read from. */
readseg = sample / sps;
sampleoff = (sample % sps);
segdone = g_atomic_int_get (&rbuf->segdone) - rbuf->segbase;
diff = readseg - segdone;
/* we can read now */
readseg = readseg % segtotal;
sampleslen = MIN (sps - sampleoff, len);
GST_DEBUG_OBJECT (ringbuffer,
"read @%p seg %d, off %d, sampleslen %d, diff %d",
dest + readseg * segsize, readseg, sampleoff, sampleslen, diff);
memcpy (data, dest + (readseg * segsize) + (sampleoff * bps),
(sampleslen * bps));
if (diff > 0)
gst_ring_buffer_advance (rbuf, diff);
len -= sampleslen;
sample += sampleslen;
data += sampleslen * bps;
}
ringbuffer->src_segment.last_stop += length;
ret = GST_FLOW_OK;
}
gst_object_unref (ringbuffer);
return ret;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (ringbuffer, "we are flushing");
GST_OBJECT_UNLOCK (ringbuffer);
gst_object_unref (ringbuffer);
return GST_FLOW_FLUSHING;
}
wrong_size:
{
GST_DEBUG_OBJECT (ringbuffer, "wrong size");
GST_ELEMENT_ERROR (ringbuffer, STREAM, WRONG_TYPE,
(NULL), ("asked to pull buffer of wrong size."));
return GST_FLOW_ERROR;
}
}
static gboolean
gst_audio_ringbuffer_src_checkgetrange_function (GstPad * pad)
{
gboolean ret;
/* we can always operate in pull mode */
ret = TRUE;
return ret;
}
/* sink currently only operates in push mode */
static gboolean
gst_audio_ringbuffer_sink_activate_push (GstPad * pad, gboolean active)
{
gboolean result = TRUE;
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad));
if (active) {
GST_DEBUG_OBJECT (ringbuffer, "activating push mode");
ringbuffer->is_eos = FALSE;
ringbuffer->pulling = FALSE;
} else {
/* unblock chain function */
GST_DEBUG_OBJECT (ringbuffer, "deactivating push mode");
ringbuffer->pulling = FALSE;
}
gst_object_unref (ringbuffer);
return result;
}
/* src operating in push mode, we will proxy the push from upstream, basically
* acting as a passthrough element. */
static gboolean
gst_audio_ringbuffer_src_activate_push (GstPad * pad, gboolean active)
{
gboolean result = FALSE;
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad));
if (active) {
GST_DEBUG_OBJECT (ringbuffer, "activating push mode");
ringbuffer->is_eos = FALSE;
ringbuffer->pushing = TRUE;
ringbuffer->pulling = FALSE;
result = TRUE;
} else {
GST_DEBUG_OBJECT (ringbuffer, "deactivating push mode");
ringbuffer->pushing = FALSE;
ringbuffer->pulling = FALSE;
result = TRUE;
}
gst_object_unref (ringbuffer);
return result;
}
/* pull mode, downstream will call our getrange function */
static gboolean
gst_audio_ringbuffer_src_activate_pull (GstPad * pad, gboolean active)
{
gboolean result;
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad));
if (active) {
GST_DEBUG_OBJECT (ringbuffer, "activating pull mode");
/* try to activate upstream in pull mode as well. If it fails, no problems,
* we'll be activated in push mode. Remember that we are pulling-through */
ringbuffer->pulling = gst_pad_activate_pull (ringbuffer->sinkpad, active);
ringbuffer->is_eos = FALSE;
ringbuffer->waiting = TRUE;
ringbuffer->flushing = FALSE;
gst_segment_init (&ringbuffer->src_segment, GST_FORMAT_BYTES);
result = TRUE;
} else {
GST_DEBUG_OBJECT (ringbuffer, "deactivating pull mode");
if (ringbuffer->pulling)
gst_pad_activate_pull (ringbuffer->sinkpad, active);
ringbuffer->pulling = FALSE;
ringbuffer->waiting = FALSE;
ringbuffer->flushing = TRUE;
result = TRUE;
}
gst_object_unref (ringbuffer);
return result;
}
static GstStateChangeReturn
gst_audio_ringbuffer_change_state (GstElement * element,
GstStateChange transition)
{
GstAudioRingbuffer *ringbuffer;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
ringbuffer = GST_AUDIO_RINGBUFFER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (ringbuffer->buffer == NULL) {
ringbuffer->buffer = gst_int_ring_buffer_new ();
gst_object_set_parent (GST_OBJECT (ringbuffer->buffer),
GST_OBJECT (ringbuffer));
gst_ring_buffer_open_device (ringbuffer->buffer);
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ringbuffer->next_sample = -1;
ringbuffer->last_align = -1;
gst_ring_buffer_set_flushing (ringbuffer->buffer, FALSE);
gst_ring_buffer_may_start (ringbuffer->buffer, TRUE);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_OBJECT_LOCK (ringbuffer);
ringbuffer->flushing = TRUE;
ringbuffer->waiting = FALSE;
g_cond_broadcast (ringbuffer->cond);
GST_OBJECT_UNLOCK (ringbuffer);
gst_ring_buffer_set_flushing (ringbuffer->buffer, TRUE);
gst_ring_buffer_may_start (ringbuffer->buffer, FALSE);
break;
default:
break;
}
ret =
GST_ELEMENT_CLASS (elem_parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_activate (ringbuffer->buffer, FALSE);
gst_ring_buffer_release (ringbuffer->buffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (ringbuffer->buffer != NULL) {
gst_ring_buffer_close_device (ringbuffer->buffer);
gst_object_unparent (GST_OBJECT (ringbuffer->buffer));
ringbuffer->buffer = NULL;
}
break;
default:
break;
}
return ret;
}
static void
gst_audio_ringbuffer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
ringbuffer->buffer_time = g_value_get_int64 (value);
break;
case PROP_SEGMENT_TIME:
ringbuffer->segment_time = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_ringbuffer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstAudioRingbuffer *ringbuffer;
ringbuffer = GST_AUDIO_RINGBUFFER (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, ringbuffer->buffer_time);
break;
case PROP_SEGMENT_TIME:
g_value_set_int64 (value, ringbuffer->segment_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audioringbuffer_debug, "audioringbuffer", 0,
"Audio ringbuffer element");
#ifdef ENABLE_NLS
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
LOCALEDIR);
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif /* ENABLE_NLS */
return gst_element_register (plugin, "audioringbuffer", GST_RANK_NONE,
GST_TYPE_AUDIO_RINGBUFFER);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audioringbuffer,
"An audio ringbuffer", plugin_init, VERSION, GST_LICENSE,
GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)