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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1181 lines
34 KiB
C
1181 lines
34 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>
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*
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* gstaudioringbuffer.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioringbuffer
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* @short_description: Asynchronous audio ringbuffer.
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*
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <glib/gstdio.h>
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#include <gst/gst.h>
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#include <gst/gst-i18n-plugin.h>
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#include <gst/audio/gstringbuffer.h>
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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GST_DEBUG_CATEGORY_STATIC (audioringbuffer_debug);
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#define GST_CAT_DEFAULT (audioringbuffer_debug)
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enum
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{
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LAST_SIGNAL
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};
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#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
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#define DEFAULT_SEGMENT_TIME ((10 * GST_MSECOND) / GST_USECOND)
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enum
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{
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PROP_0,
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PROP_BUFFER_TIME,
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PROP_SEGMENT_TIME,
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PROP_LAST
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};
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#define GST_TYPE_AUDIO_RINGBUFFER \
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(gst_audio_ringbuffer_get_type())
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#define GST_AUDIO_RINGBUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbuffer))
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#define GST_AUDIO_RINGBUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbufferClass))
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#define GST_IS_AUDIO_RINGBUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RINGBUFFER))
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#define GST_IS_AUDIO_RINGBUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RINGBUFFER))
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#define GST_AUDIO_RINGBUFFER_CAST(obj) \
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((GstAudioRingbuffer *)(obj))
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static GType gst_audio_ringbuffer_get_type (void);
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typedef struct _GstAudioRingbuffer GstAudioRingbuffer;
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typedef struct _GstAudioRingbufferClass GstAudioRingbufferClass;
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typedef struct _GstIntRingBuffer GstIntRingBuffer;
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typedef struct _GstIntRingBufferClass GstIntRingBufferClass;
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struct _GstAudioRingbuffer
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{
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GstElement element;
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/*< private > */
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GstPad *sinkpad;
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GstPad *srcpad;
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gboolean pushing;
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gboolean pulling;
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/* segments to keep track of timestamps */
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GstSegment sink_segment;
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GstSegment src_segment;
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/* flowreturn when srcpad is paused */
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gboolean is_eos;
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gboolean flushing;
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gboolean waiting;
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GCond *cond;
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GstRingBuffer *buffer;
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GstClockTime buffer_time;
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GstClockTime segment_time;
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guint64 next_sample;
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guint64 last_align;
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};
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struct _GstAudioRingbufferClass
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{
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GstElementClass parent_class;
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};
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#define GST_TYPE_INT_RING_BUFFER (gst_int_ring_buffer_get_type())
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#define GST_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_INT_RING_BUFFER,GstIntRingBuffer))
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#define GST_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_INT_RING_BUFFER,GstIntRingBufferClass))
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#define GST_INT_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_INT_RING_BUFFER, GstIntRingBufferClass))
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#define GST_INT_RING_BUFFER_CAST(obj) ((GstIntRingBuffer *)obj)
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#define GST_IS_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_INT_RING_BUFFER))
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#define GST_IS_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_INT_RING_BUFFER))
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struct _GstIntRingBuffer
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{
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GstRingBuffer object;
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};
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struct _GstIntRingBufferClass
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{
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GstRingBufferClass parent_class;
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};
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GST_BOILERPLATE (GstIntRingBuffer, gst_int_ring_buffer, GstRingBuffer,
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GST_TYPE_RING_BUFFER);
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static gboolean
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gst_int_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
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{
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spec->seglatency = spec->segtotal;
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buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
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memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
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return TRUE;
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}
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static gboolean
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gst_int_ring_buffer_release (GstRingBuffer * buf)
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{
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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return TRUE;
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}
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static gboolean
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gst_int_ring_buffer_start (GstRingBuffer * buf)
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{
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GstAudioRingbuffer *ringbuffer;
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ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (buf));
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GST_OBJECT_LOCK (ringbuffer);
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if (G_UNLIKELY (ringbuffer->waiting)) {
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ringbuffer->waiting = FALSE;
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GST_DEBUG_OBJECT (ringbuffer, "start, sending signal");
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g_cond_broadcast (ringbuffer->cond);
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}
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GST_OBJECT_UNLOCK (ringbuffer);
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return TRUE;
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}
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static void
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gst_int_ring_buffer_base_init (gpointer klass)
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{
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}
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static void
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gst_int_ring_buffer_class_init (GstIntRingBufferClass * klass)
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{
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GstRingBufferClass *gstringbuffer_class;
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gstringbuffer_class = (GstRingBufferClass *) klass;
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_int_ring_buffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_int_ring_buffer_release);
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gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_int_ring_buffer_start);
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}
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static void
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gst_int_ring_buffer_init (GstIntRingBuffer * buff,
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GstIntRingBufferClass * g_class)
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{
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}
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static GstRingBuffer *
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gst_int_ring_buffer_new (void)
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{
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GstRingBuffer *res;
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res = g_object_new (GST_TYPE_INT_RING_BUFFER, NULL);
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return res;
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}
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/* can't use boilerplate as we need to register with Queue2 to avoid conflicts
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* with ringbuffer in core elements */
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static void gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass);
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static void gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer,
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GstAudioRingbufferClass * g_class);
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static GstElementClass *elem_parent_class;
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static GType
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gst_audio_ringbuffer_get_type (void)
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{
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static GType gst_audio_ringbuffer_type = 0;
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if (!gst_audio_ringbuffer_type) {
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static const GTypeInfo gst_audio_ringbuffer_info = {
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sizeof (GstAudioRingbufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_audio_ringbuffer_class_init,
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NULL,
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NULL,
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sizeof (GstAudioRingbuffer),
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0,
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(GInstanceInitFunc) gst_audio_ringbuffer_init,
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NULL
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};
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gst_audio_ringbuffer_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstAudioRingbuffer",
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&gst_audio_ringbuffer_info, 0);
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}
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return gst_audio_ringbuffer_type;
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}
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static void gst_audio_ringbuffer_finalize (GObject * object);
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static void gst_audio_ringbuffer_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audio_ringbuffer_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_audio_ringbuffer_chain (GstPad * pad,
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GstBuffer * buffer);
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static GstFlowReturn gst_audio_ringbuffer_bufferalloc (GstPad * pad,
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guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
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static gboolean gst_audio_ringbuffer_handle_sink_event (GstPad * pad,
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GstEvent * event);
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static gboolean gst_audio_ringbuffer_handle_src_event (GstPad * pad,
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GstEvent * event);
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static gboolean gst_audio_ringbuffer_handle_src_query (GstPad * pad,
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GstQuery * query);
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static GstCaps *gst_audio_ringbuffer_getcaps (GstPad * pad);
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static gboolean gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_audio_ringbuffer_get_range (GstPad * pad,
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guint64 offset, guint length, GstBuffer ** buffer);
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static gboolean gst_audio_ringbuffer_src_checkgetrange_function (GstPad * pad);
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static gboolean gst_audio_ringbuffer_src_activate_pull (GstPad * pad,
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gboolean active);
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static gboolean gst_audio_ringbuffer_src_activate_push (GstPad * pad,
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gboolean active);
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static gboolean gst_audio_ringbuffer_sink_activate_push (GstPad * pad,
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gboolean active);
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static GstStateChangeReturn gst_audio_ringbuffer_change_state (GstElement *
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element, GstStateChange transition);
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/* static guint gst_audio_ringbuffer_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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elem_parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_property);
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g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
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g_param_spec_int64 ("buffer-time", "Buffer Time",
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"Size of audio buffer in nanoseconds", 1,
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G_MAXINT64, DEFAULT_BUFFER_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SEGMENT_TIME,
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g_param_spec_int64 ("segment-time", "Segment Time",
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"Audio segment duration in nanoseconds", 1,
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G_MAXINT64, DEFAULT_SEGMENT_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sinktemplate));
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gst_element_class_set_details_simple (gstelement_class, "AudioRingbuffer",
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"Generic",
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"Asynchronous Audio ringbuffer", "Wim Taymans <wim.taymans@gmail.com>");
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/* set several parent class virtual functions */
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_finalize);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_change_state);
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}
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static void
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gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer,
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GstAudioRingbufferClass * g_class)
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{
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ringbuffer->sinkpad =
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gst_pad_new_from_static_template (&sinktemplate, "sink");
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gst_pad_set_chain_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_chain));
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gst_pad_set_activatepush_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_sink_activate_push));
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gst_pad_set_event_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_sink_event));
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gst_pad_set_getcaps_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps));
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gst_pad_set_setcaps_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_setcaps));
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gst_pad_set_bufferalloc_function (ringbuffer->sinkpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_bufferalloc));
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gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->sinkpad);
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ringbuffer->srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
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gst_pad_set_activatepull_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_pull));
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gst_pad_set_activatepush_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_push));
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gst_pad_set_getrange_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_range));
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gst_pad_set_checkgetrange_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_checkgetrange_function));
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gst_pad_set_getcaps_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps));
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gst_pad_set_event_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_event));
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gst_pad_set_query_function (ringbuffer->srcpad,
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GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_query));
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gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->srcpad);
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gst_segment_init (&ringbuffer->sink_segment, GST_FORMAT_TIME);
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ringbuffer->cond = g_cond_new ();
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ringbuffer->is_eos = FALSE;
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ringbuffer->buffer_time = DEFAULT_BUFFER_TIME;
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ringbuffer->segment_time = DEFAULT_SEGMENT_TIME;
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GST_DEBUG_OBJECT (ringbuffer,
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"initialized ringbuffer's not_empty & not_full conditions");
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}
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/* called only once, as opposed to dispose */
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static void
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gst_audio_ringbuffer_finalize (GObject * object)
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{
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GstAudioRingbuffer *ringbuffer = GST_AUDIO_RINGBUFFER (object);
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GST_DEBUG_OBJECT (ringbuffer, "finalizing ringbuffer");
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g_cond_free (ringbuffer->cond);
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G_OBJECT_CLASS (elem_parent_class)->finalize (object);
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}
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static GstCaps *
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gst_audio_ringbuffer_getcaps (GstPad * pad)
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{
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GstAudioRingbuffer *ringbuffer;
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GstPad *otherpad;
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GstCaps *result;
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ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
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otherpad =
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(pad == ringbuffer->srcpad ? ringbuffer->sinkpad : ringbuffer->srcpad);
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result = gst_pad_peer_get_caps (otherpad);
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if (result == NULL)
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result = gst_caps_new_any ();
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return result;
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}
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static gboolean
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gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstAudioRingbuffer *ringbuffer;
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GstRingBufferSpec *spec;
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ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad));
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if (!ringbuffer->buffer)
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return FALSE;
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spec = &ringbuffer->buffer->spec;
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GST_DEBUG_OBJECT (ringbuffer, "release old ringbuffer");
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/* release old ringbuffer */
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gst_ring_buffer_activate (ringbuffer->buffer, FALSE);
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gst_ring_buffer_release (ringbuffer->buffer);
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GST_DEBUG_OBJECT (ringbuffer, "parse caps");
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spec->buffer_time = ringbuffer->buffer_time;
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spec->latency_time = ringbuffer->segment_time;
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/* parse new caps */
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if (!gst_ring_buffer_parse_caps (spec, caps))
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goto parse_error;
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gst_ring_buffer_debug_spec_buff (spec);
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GST_DEBUG_OBJECT (ringbuffer, "acquire ringbuffer");
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if (!gst_ring_buffer_acquire (ringbuffer->buffer, spec))
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goto acquire_error;
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GST_DEBUG_OBJECT (ringbuffer, "activate ringbuffer");
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gst_ring_buffer_activate (ringbuffer->buffer, TRUE);
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/* calculate actual latency and buffer times.
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* FIXME: In 0.11, store the latency_time internally in ns */
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spec->latency_time = gst_util_uint64_scale (spec->segsize,
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(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
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spec->buffer_time = spec->segtotal * spec->latency_time;
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gst_ring_buffer_debug_spec_buff (spec);
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return TRUE;
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/* ERRORS */
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parse_error:
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{
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GST_DEBUG_OBJECT (ringbuffer, "could not parse caps");
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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)
|