gstreamer/gst-libs/gst/audio/gstbaseaudiosrc.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosrc.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 "gstbaseaudiosrc.h"
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
#define GST_CAT_DEFAULT gst_base_audio_src_debug
/* BaseAudioSrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#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_src_debug, "baseaudiosrc", 0, "baseaudiosrc element");
GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_base_audio_src, GstPushSrc,
GST_TYPE_PUSH_SRC, _do_init);
static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_base_audio_src_fixate (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
element, GstStateChange transition);
static GstClock *gst_base_audio_src_provide_clock (GstElement * elem);
static GstClockTime gst_base_audio_src_get_time (GstClock * clock,
GstBaseAudioSrc * src);
static GstFlowReturn gst_base_audio_src_create (GstPushSrc * psrc,
GstBuffer ** buf);
static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
//static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 };
static void
gst_base_audio_src_base_init (gpointer g_class)
{
}
static void
gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
{
gchar *longdesc;
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpushsrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstpushsrc_class = (GstPushSrcClass *) klass;
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_base_audio_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
longdesc =
g_strdup_printf
("Size of audio buffer in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " us)", DEFAULT_BUFFER_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time", longdesc, -1,
G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
g_free (longdesc);
longdesc =
g_strdup_printf ("Audio latency in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " us)", DEFAULT_LATENCY_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time", longdesc, -1,
G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
g_free (longdesc);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock);
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
}
static void
gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
GstBaseAudioSrcClass * g_class)
{
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosrc->clock = gst_audio_clock_new ("clock",
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
gst_pad_set_fixatecaps_function (GST_BASE_SRC_PAD (baseaudiosrc),
gst_base_audio_src_fixate);
}
static GstClock *
gst_base_audio_src_provide_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (elem);
return GST_CLOCK (gst_object_ref (GST_OBJECT (src->clock)));
}
static GstClockTime
gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
{
guint64 samples;
GstClockTime result;
if (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0)
return 0;
samples = gst_ring_buffer_samples_done (src->ringbuffer);
result = samples * GST_SECOND / src->ringbuffer->spec.rate;
return result;
}
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
src->buffer_time = g_value_get_int64 (value);
break;
case PROP_LATENCY_TIME:
src->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_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, src->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_src_fixate (GstPad * pad, GstCaps * caps)
{
GstStructure *s;
s = gst_caps_get_structure (caps, 0);
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, "depth", 16);
gst_structure_fixate_field_nearest_int (s, "width", 16);
gst_structure_set (s, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
if (gst_structure_has_field (s, "endianness"))
gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
}
static gboolean
gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstRingBufferSpec *spec;
spec = &src->ringbuffer->spec;
spec->buffer_time = src->buffer_time;
spec->latency_time = src->latency_time;
if (!gst_ring_buffer_parse_caps (spec, caps))
goto parse_error;
/* calculate suggested segsize and segtotal */
spec->segsize =
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
spec->segtotal = spec->buffer_time / spec->latency_time;
GST_DEBUG ("release old ringbuffer");
gst_ring_buffer_release (src->ringbuffer);
gst_ring_buffer_debug_spec_buff (spec);
GST_DEBUG ("acquire new ringbuffer");
if (!gst_ring_buffer_acquire (src->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_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* ne need to sync to a clock here, we schedule the samples based
* on our own clock for the moment. FIXME, implement this when
* we are not using our own clock */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
gst_ring_buffer_pause (src->ringbuffer);
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
src->next_sample = -1;
gst_ring_buffer_clear_all (src->ringbuffer);
break;
default:
break;
}
return TRUE;
}
static GstFlowReturn
gst_base_audio_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (psrc);
GstBuffer *buf;
guchar *data;
guint len, samples;
guint res;
guint64 sample;
if (!gst_ring_buffer_is_acquired (src->ringbuffer))
goto wrong_state;
buf = gst_buffer_new_and_alloc (src->ringbuffer->spec.segsize);
data = GST_BUFFER_DATA (buf);
len = GST_BUFFER_SIZE (buf);
if (src->next_sample != -1) {
sample = src->next_sample;
} else {
sample = 0;
}
samples = len / src->ringbuffer->spec.bytes_per_sample;
res = gst_ring_buffer_read (src->ringbuffer, sample, data, samples);
if (res == -1)
goto stopped;
src->next_sample = sample + samples;
gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (psrc)));
*outbuf = buf;
return GST_FLOW_OK;
wrong_state:
{
GST_DEBUG ("ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG ("ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
GstRingBuffer *
gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstBaseAudioSrcClass *bclass;
GstRingBuffer *buffer = NULL;
bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (src);
if (buffer) {
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (src));
}
return buffer;
}
void
gst_base_audio_src_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
gpointer user_data)
{
//GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (data);
}
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (src->ringbuffer == NULL) {
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
gst_ring_buffer_set_callback (src->ringbuffer,
gst_base_audio_src_callback, src);
}
if (!gst_ring_buffer_open_device (src->ringbuffer))
return GST_STATE_CHANGE_FAILURE;
src->next_sample = 0;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
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 (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
gst_ring_buffer_release (src->ringbuffer);
src->next_sample = 0;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_ring_buffer_close_device (src->ringbuffer);
gst_object_unref (src->ringbuffer);
src->ringbuffer = NULL;
break;
default:
break;
}
return ret;
}