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ac1cc82165
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_drain): Our EOS time contains the base_time, _wait_eos() expects a running_time so we have to subtract the base_time again before calling the function. This fixes an EOS regression where the base_time was added twice and EOS took longer and longer in certain situations. Fixes #498767.
1528 lines
47 KiB
C
1528 lines
47 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosink.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:gstbaseaudiosink
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* @short_description: Base class for audio sinks
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* @see_also: #GstAudioSink, #GstRingBuffer.
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*
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* This is the base class for audio sinks. Subclasses need to implement the
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* ::create_ringbuffer vmethod. This base class will then take care of
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* writing samples to the ringbuffer, synchronisation, clipping and flushing.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#include <string.h>
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#include "gstbaseaudiosink.h"
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GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug);
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#define GST_CAT_DEFAULT gst_base_audio_sink_debug
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#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate))
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struct _GstBaseAudioSinkPrivate
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{
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/* upstream latency */
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GstClockTime us_latency;
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/* the clock slaving algorithm in use */
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GstBaseAudioSinkSlaveMethod slave_method;
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/* running average of clock skew */
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GstClockTimeDiff avg_skew;
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/* the number of samples we aligned last time */
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gint64 last_align;
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};
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/* BaseAudioSink signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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/* we tollerate half a second diff before we start resyncing. This
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* should be enough to compensate for various rounding errors in the timestamp
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* and sample offset position.
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* This is an emergency resync fallback since buffers marked as DISCONT will
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* always lock to the correct timestamp immediatly and buffers not marked as
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* DISCONT are contiguous by definition.
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*/
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#define DIFF_TOLERANCE 2
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/* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
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#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
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#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
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#define DEFAULT_PROVIDE_CLOCK TRUE
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#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SINK_SLAVE_SKEW
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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_LATENCY_TIME,
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PROP_PROVIDE_CLOCK,
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PROP_SLAVE_METHOD
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};
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#define GST_TYPE_SLAVE_METHOD (slave_method_get_type ())
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static GType
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slave_method_get_type (void)
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{
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static GType slave_method_type = 0;
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static const GEnumValue slave_method[] = {
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{GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
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{GST_BASE_AUDIO_SINK_SLAVE_SKEW, "Skew slaving", "skew"},
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{GST_BASE_AUDIO_SINK_SLAVE_NONE, "No slaving", "none"},
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{0, NULL, NULL},
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};
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if (!slave_method_type) {
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slave_method_type =
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g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
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}
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return slave_method_type;
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}
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
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GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_base_audio_sink, GstBaseSink,
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GST_TYPE_BASE_SINK, _do_init);
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static void gst_base_audio_sink_dispose (GObject * object);
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static void gst_base_audio_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_base_audio_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
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basesink);
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static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
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gboolean active);
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static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
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query);
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static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
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static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
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GstBaseAudioSink * sink);
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static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
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guint len, gpointer user_data);
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static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink,
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GstBuffer * buffer);
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static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink,
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GstBuffer * buffer);
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static gboolean gst_base_audio_sink_event (GstBaseSink * bsink,
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GstEvent * event);
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static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
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GstCaps * caps);
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static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
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/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_base_audio_sink_base_init (gpointer g_class)
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{
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}
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static void
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gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate));
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_sink_dispose);
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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 microseconds", 1,
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G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
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g_param_spec_int64 ("latency-time", "Latency Time",
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"Audio latency in microseconds", 1,
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G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
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g_param_spec_boolean ("provide-clock", "Provide Clock",
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"Provide a clock to be used as the global pipeline clock",
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DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
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g_param_spec_enum ("slave-method", "Slave Method",
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"Algorithm to use to match the rate of the masterclock",
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GST_TYPE_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, G_PARAM_READWRITE));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
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gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
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gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
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gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
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gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
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gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render);
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gstbasesink_class->get_times =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
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gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
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gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate);
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gstbasesink_class->async_play =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
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gstbasesink_class->activate_pull =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull);
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}
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static void
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gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
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GstBaseAudioSinkClass * g_class)
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{
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baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
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baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
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baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
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baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
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baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD;
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baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
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GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
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/* FIXME, enable pull mode when segments, latency, state changes, negotiation
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* and clock slaving are figured out */
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GST_BASE_SINK (baseaudiosink)->can_activate_pull = FALSE;
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}
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static void
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gst_base_audio_sink_dispose (GObject * object)
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{
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GstBaseAudioSink *sink;
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sink = GST_BASE_AUDIO_SINK (object);
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if (sink->provided_clock)
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gst_object_unref (sink->provided_clock);
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sink->provided_clock = NULL;
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if (sink->ringbuffer) {
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gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
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sink->ringbuffer = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static GstClock *
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gst_base_audio_sink_provide_clock (GstElement * elem)
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{
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GstBaseAudioSink *sink;
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GstClock *clock;
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sink = GST_BASE_AUDIO_SINK (elem);
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/* we have no ringbuffer (must be NULL state) */
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if (sink->ringbuffer == NULL)
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goto wrong_state;
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if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
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goto wrong_state;
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GST_OBJECT_LOCK (sink);
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if (!sink->provide_clock)
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goto clock_disabled;
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clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
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GST_OBJECT_UNLOCK (sink);
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return clock;
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/* ERRORS */
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wrong_state:
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{
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GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
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return NULL;
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}
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clock_disabled:
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{
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GST_DEBUG_OBJECT (sink, "clock provide disabled");
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GST_OBJECT_UNLOCK (sink);
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return NULL;
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}
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}
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static gboolean
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gst_base_audio_sink_query (GstElement * element, GstQuery * query)
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{
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gboolean res = FALSE;
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GstBaseAudioSink *basesink = GST_BASE_AUDIO_SINK (element);
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:
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{
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gboolean live, us_live;
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GstClockTime min_l, max_l;
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GST_DEBUG_OBJECT (basesink, "latency query");
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if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) {
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GST_DEBUG_OBJECT (basesink,
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"we are not yet negotiated, can't report latency yet");
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res = FALSE;
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goto done;
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}
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/* ask parent first, it will do an upstream query for us. */
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if ((res =
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gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
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&us_live, &min_l, &max_l))) {
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GstClockTime min_latency, max_latency;
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/* we and upstream are both live, adjust the min_latency */
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if (live && us_live) {
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GstRingBufferSpec *spec;
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spec = &basesink->ringbuffer->spec;
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basesink->priv->us_latency = min_l;
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min_latency =
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gst_util_uint64_scale_int (spec->segtotal * spec->segsize,
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GST_SECOND, spec->rate * spec->bytes_per_sample);
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/* we cannot go lower than the buffer size and the min peer latency */
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min_latency = min_latency + min_l;
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/* the max latency is the max of the peer, we can delay an infinite
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* amount of time. */
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max_latency = min_latency + (max_l == -1 ? 0 : max_l);
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GST_DEBUG_OBJECT (basesink,
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"peer min %" GST_TIME_FORMAT ", our min latency: %"
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GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
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GST_TIME_ARGS (min_latency));
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} else {
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GST_DEBUG_OBJECT (basesink,
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"peer or we are not live, don't care about latency");
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min_latency = 0;
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max_latency = -1;
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}
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gst_query_set_latency (query, live, min_latency, max_latency);
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}
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break;
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}
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default:
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res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
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break;
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}
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done:
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return res;
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}
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static GstClockTime
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gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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{
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guint64 raw, samples;
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guint delay;
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GstClockTime result, us_latency;
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if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
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return GST_CLOCK_TIME_NONE;
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/* our processed samples are always increasing */
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raw = samples = gst_ring_buffer_samples_done (sink->ringbuffer);
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/* the number of samples not yet processed, this is still queued in the
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* device (not played for playback). */
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delay = gst_ring_buffer_delay (sink->ringbuffer);
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if (G_LIKELY (samples >= delay))
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samples -= delay;
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else
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samples = 0;
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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sink->ringbuffer->spec.rate);
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/* latency before starting the clock */
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us_latency = sink->priv->us_latency;
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result += us_latency;
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GST_DEBUG_OBJECT (sink,
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"processed samples: raw %llu, delay %u, real %llu, time %"
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GST_TIME_FORMAT ", upstream latency %" GST_TIME_FORMAT, raw, delay,
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samples, GST_TIME_ARGS (result), GST_TIME_ARGS (us_latency));
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return result;
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}
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/**
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* gst_base_audio_sink_set_provide_clock:
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* @sink: a #GstBaseAudioSink
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* @provide: new state
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*
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* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
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* gst_element_provide_clock() will return a clock that reflects the datarate
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* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
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*
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* Since: 0.10.16
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*/
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void
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gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink,
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gboolean provide)
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{
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g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
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GST_OBJECT_LOCK (sink);
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sink->provide_clock = provide;
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GST_OBJECT_UNLOCK (sink);
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}
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/**
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* gst_base_audio_sink_get_provide_clock:
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* @sink: a #GstBaseAudioSink
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*
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* Queries whether @sink will provide a clock or not. See also
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* gst_base_audio_sink_set_provide_clock.
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*
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* Returns: %TRUE if @sink will provide a clock.
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*
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* Since: 0.10.16
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*/
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gboolean
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gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink)
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{
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gboolean result;
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g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE);
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GST_OBJECT_LOCK (sink);
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result = sink->provide_clock;
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GST_OBJECT_UNLOCK (sink);
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return result;
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}
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/**
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* gst_base_audio_sink_set_slave_method:
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* @sink: a #GstBaseAudioSink
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* @method: the new slave method
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*
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* Controls how clock slaving will be performed in @sink.
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*
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* Since: 0.10.16
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*/
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void
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gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink,
|
|
GstBaseAudioSinkSlaveMethod method)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->slave_method = method;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_audio_sink_get_slave_method:
|
|
* @sink: a #GstBaseAudioSink
|
|
*
|
|
* Get the current slave method used by @sink.
|
|
*
|
|
* Returns: The current slave method used by @sink.
|
|
*
|
|
* Since: 0.10.16
|
|
*/
|
|
GstBaseAudioSinkSlaveMethod
|
|
gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink)
|
|
{
|
|
GstBaseAudioSinkSlaveMethod result;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = sink->priv->slave_method;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioSink *sink;
|
|
|
|
sink = GST_BASE_AUDIO_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BUFFER_TIME:
|
|
sink->buffer_time = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_LATENCY_TIME:
|
|
sink->latency_time = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_PROVIDE_CLOCK:
|
|
gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_SLAVE_METHOD:
|
|
gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_sink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioSink *sink;
|
|
|
|
sink = GST_BASE_AUDIO_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BUFFER_TIME:
|
|
g_value_set_int64 (value, sink->buffer_time);
|
|
break;
|
|
case PROP_LATENCY_TIME:
|
|
g_value_set_int64 (value, sink->latency_time);
|
|
break;
|
|
case PROP_PROVIDE_CLOCK:
|
|
g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink));
|
|
break;
|
|
case PROP_SLAVE_METHOD:
|
|
g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
|
|
GstRingBufferSpec *spec;
|
|
|
|
if (!sink->ringbuffer)
|
|
return FALSE;
|
|
|
|
spec = &sink->ringbuffer->spec;
|
|
|
|
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
|
|
|
|
/* release old ringbuffer */
|
|
gst_ring_buffer_release (sink->ringbuffer);
|
|
|
|
GST_DEBUG_OBJECT (sink, "parse caps");
|
|
|
|
spec->buffer_time = sink->buffer_time;
|
|
spec->latency_time = sink->latency_time;
|
|
|
|
/* parse new caps */
|
|
if (!gst_ring_buffer_parse_caps (spec, caps))
|
|
goto parse_error;
|
|
|
|
gst_ring_buffer_debug_spec_buff (spec);
|
|
|
|
GST_DEBUG_OBJECT (sink, "acquire new ringbuffer");
|
|
|
|
if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
|
|
goto acquire_error;
|
|
|
|
/* calculate actual latency and buffer times.
|
|
* FIXME: In 0.11, store the latency_time internally in ns */
|
|
spec->latency_time = gst_util_uint64_scale (spec->segsize,
|
|
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
|
|
|
|
spec->buffer_time = spec->segtotal * spec->latency_time;
|
|
|
|
gst_ring_buffer_debug_spec_buff (spec);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
parse_error:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "could not parse caps");
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
|
|
(NULL), ("cannot parse audio format."));
|
|
return FALSE;
|
|
}
|
|
acquire_error:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstStructure *s;
|
|
gint width, depth;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
/* fields for all formats */
|
|
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, "width", 16);
|
|
|
|
/* fields for int */
|
|
if (gst_structure_has_field (s, "depth")) {
|
|
gst_structure_get_int (s, "width", &width);
|
|
/* round width to nearest multiple of 8 for the depth */
|
|
depth = GST_ROUND_UP_8 (width);
|
|
gst_structure_fixate_field_nearest_int (s, "depth", depth);
|
|
}
|
|
if (gst_structure_has_field (s, "signed"))
|
|
gst_structure_fixate_field_boolean (s, "signed", TRUE);
|
|
if (gst_structure_has_field (s, "endianness"))
|
|
gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
/* our clock sync is a bit too much for the base class to handle so
|
|
* we implement it ourselves. */
|
|
*start = GST_CLOCK_TIME_NONE;
|
|
*end = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
/* This waits for the drain to happen and can be canceled */
|
|
static gboolean
|
|
gst_base_audio_sink_drain (GstBaseAudioSink * sink)
|
|
{
|
|
GstClockTime base_time;
|
|
|
|
if (!sink->ringbuffer)
|
|
return TRUE;
|
|
if (!sink->ringbuffer->spec.rate)
|
|
return TRUE;
|
|
|
|
/* need to start playback before we can drain, but only when
|
|
* we have successfully negotiated a format and thus acquired the
|
|
* ringbuffer. */
|
|
if (gst_ring_buffer_is_acquired (sink->ringbuffer))
|
|
gst_ring_buffer_start (sink->ringbuffer);
|
|
|
|
if (sink->next_sample != -1) {
|
|
GstClockTime time;
|
|
|
|
/* convert next expected sample to time */
|
|
time =
|
|
gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
|
|
sink->ringbuffer->spec.rate);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last sample %" G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
|
|
sink->next_sample, GST_TIME_ARGS (time));
|
|
|
|
/* our time already includes the base_time, _wait_eos() wants a running_time
|
|
* so we have to subtract the base_time again here. FIXME, store an
|
|
* unadjusted EOS time so that we don't have to do this. */
|
|
GST_OBJECT_LOCK (sink);
|
|
base_time = GST_ELEMENT_CAST (sink)->base_time;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
if (time > base_time)
|
|
time -= base_time;
|
|
else
|
|
time = 0;
|
|
|
|
/* wait for the EOS time to be reached, this is the time when the last
|
|
* sample is played. */
|
|
gst_base_sink_wait_eos (GST_BASE_SINK (sink), time, NULL);
|
|
|
|
sink->next_sample = -1;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
|
|
{
|
|
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
if (sink->ringbuffer)
|
|
gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* always resync on sample after a flush */
|
|
sink->priv->avg_skew = -1;
|
|
sink->next_sample = -1;
|
|
if (sink->ringbuffer)
|
|
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
|
|
break;
|
|
case GST_EVENT_EOS:
|
|
/* now wait till we played everything */
|
|
gst_base_audio_sink_drain (sink);
|
|
break;
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
gdouble rate;
|
|
|
|
/* we only need the rate */
|
|
gst_event_parse_new_segment_full (event, NULL, &rate, NULL, NULL,
|
|
NULL, NULL, NULL);
|
|
|
|
GST_DEBUG_OBJECT (sink, "new segment rate of %f", rate);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
|
|
{
|
|
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
|
|
|
|
if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
|
|
goto wrong_state;
|
|
|
|
/* we don't really do anything when prerolling. We could make a
|
|
* property to play this buffer to have some sort of scrubbing
|
|
* support. */
|
|
return GST_FLOW_OK;
|
|
|
|
wrong_state:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
}
|
|
|
|
static guint64
|
|
gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
|
|
{
|
|
guint64 sample;
|
|
gint writeseg, segdone, sps;
|
|
gint diff;
|
|
|
|
/* assume we can append to the previous sample */
|
|
sample = sink->next_sample;
|
|
/* no previous sample, try to insert at position 0 */
|
|
if (sample == -1)
|
|
sample = 0;
|
|
|
|
sps = sink->ringbuffer->samples_per_seg;
|
|
|
|
/* figure out the segment and the offset inside the segment where
|
|
* the sample should be written. */
|
|
writeseg = sample / sps;
|
|
|
|
/* get the currently processed segment */
|
|
segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
|
|
- sink->ringbuffer->segbase;
|
|
|
|
/* see how far away it is from the write segment */
|
|
diff = writeseg - segdone;
|
|
if (diff < 0) {
|
|
/* sample would be dropped, position to next playable position */
|
|
sample = (segdone + 1) * sps;
|
|
}
|
|
|
|
return sample;
|
|
}
|
|
|
|
static GstClockTime
|
|
clock_convert_external (GstClockTime external, GstClockTime cinternal,
|
|
GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom,
|
|
GstClockTime us_latency)
|
|
{
|
|
/* adjust for rate and speed */
|
|
if (external >= cexternal) {
|
|
external =
|
|
gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
|
|
external += cinternal;
|
|
} else {
|
|
external = gst_util_uint64_scale (cexternal - external,
|
|
crate_denom, crate_num);
|
|
if (cinternal > external)
|
|
external = cinternal - external;
|
|
else
|
|
external = 0;
|
|
}
|
|
/* adjust for offset when slaving started */
|
|
if (external > us_latency)
|
|
external -= us_latency;
|
|
else
|
|
external = 0;
|
|
|
|
return external;
|
|
}
|
|
|
|
/* algorithm to calculate sample positions that will result in resampling to
|
|
* match the clock rate of the master */
|
|
static void
|
|
gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
GstClockTime cinternal, cexternal;
|
|
GstClockTime crate_num, crate_denom;
|
|
|
|
/* get calibration parameters to compensate for speed and offset differences
|
|
* when we are slaved */
|
|
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
|
|
&crate_num, &crate_denom);
|
|
|
|
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
|
|
GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
|
|
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
|
|
crate_denom, gst_guint64_to_gdouble (crate_num) /
|
|
gst_guint64_to_gdouble (crate_denom));
|
|
|
|
if (crate_num == 0)
|
|
crate_denom = crate_num = 1;
|
|
|
|
/* bring external time to internal time */
|
|
render_start = clock_convert_external (render_start, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* algorithm to calculate sample positions that will result in changing the
|
|
* playout pointer to match the clock rate of the master */
|
|
static void
|
|
gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
GstClockTime cinternal, cexternal, crate_num, crate_denom;
|
|
GstClockTime etime, itime;
|
|
GstClockTimeDiff skew, segtime, segtime2;
|
|
gint segsamples;
|
|
gint64 last_align;
|
|
|
|
/* get calibration parameters to compensate for offsets */
|
|
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
|
|
&crate_num, &crate_denom);
|
|
|
|
/* sample clocks and figure out clock skew */
|
|
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
|
|
itime = gst_clock_get_internal_time (sink->provided_clock);
|
|
|
|
etime -= cexternal;
|
|
itime -= cinternal;
|
|
|
|
skew = GST_CLOCK_DIFF (etime, itime);
|
|
if (sink->priv->avg_skew == -1) {
|
|
/* first observation */
|
|
sink->priv->avg_skew = skew;
|
|
} else {
|
|
/* next observations use a moving average */
|
|
sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
|
|
GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
|
|
|
|
/* the max drift we allow is the length of a segment */
|
|
segtime = sink->ringbuffer->spec.latency_time * 1000;
|
|
segtime2 = segtime / 2;
|
|
|
|
/* adjust playout pointer based on skew */
|
|
if (sink->priv->avg_skew > segtime2) {
|
|
/* master is running slower, move internal time forward */
|
|
GST_WARNING_OBJECT (sink,
|
|
"correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
|
|
sink->priv->avg_skew, segtime2);
|
|
cinternal += segtime;
|
|
sink->priv->avg_skew -= segtime;
|
|
|
|
segsamples =
|
|
sink->ringbuffer->spec.segsize /
|
|
sink->ringbuffer->spec.bytes_per_sample;
|
|
last_align = sink->priv->last_align;
|
|
|
|
/* if we were aligning in the wrong direction or we aligned more than what we
|
|
* will correct, resync */
|
|
if (last_align < 0 || last_align > segsamples)
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " segsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, segsamples, sink->next_sample);
|
|
|
|
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
} else if (sink->priv->avg_skew < -segtime2) {
|
|
/* master is running faster, move external time forwards */
|
|
GST_WARNING_OBJECT (sink,
|
|
"correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
|
|
sink->priv->avg_skew, -segtime2);
|
|
cexternal += segtime;
|
|
sink->priv->avg_skew += segtime;
|
|
|
|
segsamples =
|
|
sink->ringbuffer->spec.segsize /
|
|
sink->ringbuffer->spec.bytes_per_sample;
|
|
last_align = sink->priv->last_align;
|
|
|
|
/* if we were aligning in the wrong direction or we aligned more than what we
|
|
* will correct, resync */
|
|
if (last_align > 0 || -last_align > segsamples)
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " segsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, segsamples, sink->next_sample);
|
|
|
|
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
}
|
|
|
|
/* convert, ignoring speed */
|
|
render_start = clock_convert_external (render_start, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* apply the clock offset but do no slaving otherwise */
|
|
static void
|
|
gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
GstClockTime cinternal, cexternal, crate_num, crate_denom;
|
|
|
|
/* get calibration parameters to compensate for offsets */
|
|
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
|
|
&crate_num, &crate_denom);
|
|
|
|
/* convert, ignoring speed */
|
|
render_start = clock_convert_external (render_start, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom, sink->priv->us_latency);
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* converts render_start and render_stop to their slaved values */
|
|
static void
|
|
gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
switch (sink->priv->slave_method) {
|
|
case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
|
|
gst_base_audio_sink_resample_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
|
|
gst_base_audio_sink_skew_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
case GST_BASE_AUDIO_SINK_SLAVE_NONE:
|
|
gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
default:
|
|
g_warning ("unknown slaving method %d", sink->priv->slave_method);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|
{
|
|
guint64 in_offset;
|
|
GstClockTime time, stop, render_start, render_stop, sample_offset;
|
|
GstBaseAudioSink *sink;
|
|
GstRingBuffer *ringbuf;
|
|
gint64 diff, align, ctime, cstop;
|
|
guint8 *data;
|
|
guint size;
|
|
guint samples, written;
|
|
gint bps;
|
|
gint accum;
|
|
gint out_samples;
|
|
GstClockTime base_time = GST_CLOCK_TIME_NONE, latency;
|
|
GstClock *clock;
|
|
gboolean sync, slaved, align_next;
|
|
|
|
sink = GST_BASE_AUDIO_SINK (bsink);
|
|
|
|
ringbuf = sink->ringbuffer;
|
|
|
|
/* can't do anything when we don't have the device */
|
|
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf)))
|
|
goto wrong_state;
|
|
|
|
bps = ringbuf->spec.bytes_per_sample;
|
|
|
|
size = GST_BUFFER_SIZE (buf);
|
|
if (G_UNLIKELY (size % bps) != 0)
|
|
goto wrong_size;
|
|
|
|
samples = size / bps;
|
|
out_samples = samples;
|
|
|
|
in_offset = GST_BUFFER_OFFSET (buf);
|
|
time = GST_BUFFER_TIMESTAMP (buf);
|
|
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
|
|
ringbuf->spec.rate);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
|
|
", samples %u", GST_TIME_ARGS (time), in_offset,
|
|
GST_TIME_ARGS (bsink->segment.start), samples);
|
|
|
|
data = GST_BUFFER_DATA (buf);
|
|
|
|
/* if not valid timestamp or we can't clip or sync, try to play
|
|
* sample ASAP */
|
|
if (!GST_CLOCK_TIME_IS_VALID (time)) {
|
|
render_start = gst_base_audio_sink_get_offset (sink);
|
|
render_stop = render_start + samples;
|
|
GST_DEBUG_OBJECT (sink,
|
|
"Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT,
|
|
GST_BUFFER_SIZE (buf), render_start);
|
|
goto no_sync;
|
|
}
|
|
|
|
/* samples should be rendered based on their timestamp. All samples
|
|
* arriving before the segment.start or after segment.stop are to be
|
|
* thrown away. All samples should also be clipped to the segment
|
|
* boundaries */
|
|
/* let's calc stop based on the number of samples in the buffer instead
|
|
* of trusting the DURATION */
|
|
if (!gst_segment_clip (&bsink->segment, GST_FORMAT_TIME, time, stop, &ctime,
|
|
&cstop))
|
|
goto out_of_segment;
|
|
|
|
/* see if some clipping happened */
|
|
diff = ctime - time;
|
|
if (diff > 0) {
|
|
/* bring clipped time to samples */
|
|
diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
|
|
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
|
|
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
|
|
samples -= diff;
|
|
data += diff * bps;
|
|
time = ctime;
|
|
}
|
|
diff = stop - cstop;
|
|
if (diff > 0) {
|
|
/* bring clipped time to samples */
|
|
diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
|
|
GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
|
|
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
|
|
samples -= diff;
|
|
stop = cstop;
|
|
}
|
|
|
|
/* figure out how to sync */
|
|
if ((clock = GST_ELEMENT_CLOCK (bsink)))
|
|
sync = bsink->sync;
|
|
else
|
|
sync = FALSE;
|
|
|
|
if (!sync) {
|
|
/* no sync needed, play sample ASAP */
|
|
render_start = gst_base_audio_sink_get_offset (sink);
|
|
render_stop = render_start + samples;
|
|
GST_DEBUG_OBJECT (sink,
|
|
"no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
|
|
goto no_sync;
|
|
}
|
|
|
|
/* bring buffer start and stop times to running time */
|
|
render_start =
|
|
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
|
|
render_stop =
|
|
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
|
|
|
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
|
|
|
|
GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (base_time));
|
|
|
|
/* add base time to sync against the clock */
|
|
render_start += base_time;
|
|
render_stop += base_time;
|
|
|
|
/* compensate for latency */
|
|
latency = gst_base_sink_get_latency (bsink);
|
|
GST_DEBUG_OBJECT (sink,
|
|
"compensating for latency %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
|
|
|
|
/* add latency to get the timestamp to sync against the pipeline clock */
|
|
render_start += latency;
|
|
render_stop += latency;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
|
|
|
if ((slaved = clock != sink->provided_clock)) {
|
|
/* handle clock slaving */
|
|
gst_base_audio_sink_handle_slaving (sink, render_start, render_stop,
|
|
&render_start, &render_stop);
|
|
}
|
|
|
|
/* and bring the time to the rate corrected offset in the buffer */
|
|
render_start = gst_util_uint64_scale_int (render_start,
|
|
ringbuf->spec.rate, GST_SECOND);
|
|
render_stop = gst_util_uint64_scale_int (render_stop,
|
|
ringbuf->spec.rate, GST_SECOND);
|
|
|
|
/* always resync after a discont */
|
|
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
|
|
GST_DEBUG_OBJECT (sink, "resync after discont");
|
|
goto no_align;
|
|
}
|
|
|
|
if (G_UNLIKELY (sink->next_sample == -1)) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"no align possible: no previous sample position known");
|
|
goto no_align;
|
|
}
|
|
|
|
/* positive playback rate, first sample is render_start, negative rate, first
|
|
* sample is render_stop */
|
|
if (bsink->segment.rate >= 1.0)
|
|
sample_offset = render_start;
|
|
else
|
|
sample_offset = render_stop;
|
|
|
|
/* now try to align the sample to the previous one */
|
|
if (sample_offset >= sink->next_sample)
|
|
diff = sample_offset - sink->next_sample;
|
|
else
|
|
diff = sink->next_sample - sample_offset;
|
|
|
|
/* we tollerate half a second diff before we start resyncing. This
|
|
* should be enough to compensate for various rounding errors in the timestamp
|
|
* and sample offset position. We always resync if we got a discont anyway and
|
|
* non-discont should be aligned by definition. */
|
|
if (G_LIKELY (diff < ringbuf->spec.rate / DIFF_TOLERANCE)) {
|
|
/* calc align with previous sample */
|
|
align = sink->next_sample - sample_offset;
|
|
GST_DEBUG_OBJECT (sink,
|
|
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align,
|
|
ringbuf->spec.rate / DIFF_TOLERANCE);
|
|
} else {
|
|
/* bring sample diff to seconds for error message */
|
|
diff = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate);
|
|
/* timestamps drifted apart from previous samples too much, we need to
|
|
* resync. We log this as an element warning. */
|
|
GST_ELEMENT_WARNING (sink, CORE, CLOCK,
|
|
("Compensating for audio synchronisation problems"),
|
|
("Unexpected discontinuity in audio timestamps of more "
|
|
"than half a second (%" GST_TIME_FORMAT "), resyncing",
|
|
GST_TIME_ARGS (diff)));
|
|
align = 0;
|
|
}
|
|
sink->priv->last_align = align;
|
|
|
|
/* apply alignment */
|
|
render_start += align;
|
|
|
|
/* only align stop if we are not slaved to resample */
|
|
if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) {
|
|
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
|
|
goto no_align;
|
|
}
|
|
render_stop += align;
|
|
|
|
no_align:
|
|
/* number of target samples is difference between start and stop */
|
|
out_samples = render_stop - render_start;
|
|
|
|
no_sync:
|
|
/* we render the first or last sample first, depending on the rate */
|
|
if (bsink->segment.rate >= 1.0)
|
|
sample_offset = render_start;
|
|
else
|
|
sample_offset = render_stop;
|
|
|
|
GST_DEBUG_OBJECT (sink, "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 (ringbuf, &sample_offset, data, samples,
|
|
out_samples, &accum);
|
|
|
|
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
|
|
/* if we wrote all, we're done */
|
|
if (written == samples)
|
|
break;
|
|
|
|
/* else something interrupted us and we wait for preroll. */
|
|
if (gst_base_sink_wait_preroll (bsink) != GST_FLOW_OK)
|
|
goto stopping;
|
|
|
|
/* 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)
|
|
sink->next_sample = sample_offset;
|
|
else
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
|
|
sink->next_sample);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"start playback because we are at the end of segment");
|
|
gst_ring_buffer_start (ringbuf);
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* SPECIAL cases */
|
|
out_of_segment:
|
|
{
|
|
GST_DEBUG_OBJECT (sink,
|
|
"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (time),
|
|
GST_TIME_ARGS (bsink->segment.start));
|
|
return GST_FLOW_OK;
|
|
}
|
|
/* ERRORS */
|
|
wrong_state:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
wrong_size:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "wrong size");
|
|
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
|
|
(NULL), ("sink received buffer of wrong size."));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer is stopping");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_audio_sink_create_ringbuffer:
|
|
* @sink: a #GstBaseAudioSink.
|
|
*
|
|
* Create and return the #GstRingBuffer for @sink. This function will call the
|
|
* ::create_ringbuffer vmethod and will set @sink as the parent of the returned
|
|
* buffer (see gst_object_set_parent()).
|
|
*
|
|
* Returns: The new ringbuffer of @sink.
|
|
*/
|
|
GstRingBuffer *
|
|
gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
|
|
{
|
|
GstBaseAudioSinkClass *bclass;
|
|
GstRingBuffer *buffer = NULL;
|
|
|
|
bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
|
|
if (bclass->create_ringbuffer)
|
|
buffer = bclass->create_ringbuffer (sink);
|
|
|
|
if (buffer)
|
|
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
|
|
|
|
return buffer;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
|
|
{
|
|
gboolean ret;
|
|
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
|
|
|
|
if (active) {
|
|
gst_ring_buffer_set_callback (sink->ringbuffer,
|
|
gst_base_audio_sink_callback, sink);
|
|
ret = gst_ring_buffer_start (sink->ringbuffer);
|
|
} else {
|
|
gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
|
|
/* stop thread */
|
|
ret = gst_ring_buffer_release (sink->ringbuffer);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
|
|
gpointer user_data)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstBaseAudioSink *sink;
|
|
GstBuffer *buf;
|
|
GstFlowReturn ret;
|
|
|
|
basesink = GST_BASE_SINK (user_data);
|
|
sink = GST_BASE_AUDIO_SINK (user_data);
|
|
|
|
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
|
|
will copy twice, once into data, once into DMA */
|
|
GST_LOG_OBJECT (basesink, "pulling %d bytes offset %" G_GUINT64_FORMAT
|
|
" to fill audio buffer", len, basesink->offset);
|
|
ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf);
|
|
|
|
if (ret != GST_FLOW_OK) {
|
|
if (ret == GST_FLOW_UNEXPECTED)
|
|
goto eos;
|
|
else
|
|
goto error;
|
|
}
|
|
|
|
if (len != GST_BUFFER_SIZE (buf)) {
|
|
GST_INFO_OBJECT (basesink, "short read pulling from sink pad: %d<%d",
|
|
len, GST_BUFFER_SIZE (buf));
|
|
len = MIN (GST_BUFFER_SIZE (buf), len);
|
|
}
|
|
|
|
basesink->offset += len;
|
|
|
|
memcpy (data, GST_BUFFER_DATA (buf), len);
|
|
|
|
return;
|
|
|
|
error:
|
|
{
|
|
GST_WARNING_OBJECT (basesink, "Got flow error but can't return it: %d",
|
|
ret);
|
|
return;
|
|
}
|
|
eos:
|
|
{
|
|
/* FIXME: this is not quite correct; we'll be called endlessly until
|
|
* the sink gets shut down; maybe we should set a flag somewhere, or
|
|
* set segment.stop and segment.duration to the last sample or so */
|
|
GST_DEBUG_OBJECT (sink, "EOS");
|
|
gst_element_post_message (GST_ELEMENT_CAST (sink),
|
|
gst_message_new_eos (GST_OBJECT_CAST (sink)));
|
|
gst_base_audio_sink_drain (sink);
|
|
}
|
|
}
|
|
|
|
/* should be called with the LOCK */
|
|
static GstStateChangeReturn
|
|
gst_base_audio_sink_async_play (GstBaseSink * basesink)
|
|
{
|
|
GstClock *clock;
|
|
GstBaseAudioSink *sink;
|
|
GstClockTime itime, etime;
|
|
GstClockTime rate_num, rate_denom;
|
|
|
|
sink = GST_BASE_AUDIO_SINK (basesink);
|
|
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
|
|
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
|
|
|
|
clock = GST_ELEMENT_CLOCK (sink);
|
|
if (clock == NULL)
|
|
goto done;
|
|
|
|
/* we provided the global clock, don't need to do anything special */
|
|
if (clock == sink->provided_clock)
|
|
goto done;
|
|
|
|
/* if we are slaved to a clock, we need to set the initial
|
|
* calibration */
|
|
/* get external and internal time to set as calibration params */
|
|
etime = gst_clock_get_time (clock);
|
|
itime = gst_clock_get_internal_time (sink->provided_clock);
|
|
|
|
sink->priv->avg_skew = -1;
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
|
|
|
|
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
|
|
&rate_denom);
|
|
gst_clock_set_calibration (sink->provided_clock, itime, etime,
|
|
rate_num, rate_denom);
|
|
|
|
switch (sink->priv->slave_method) {
|
|
case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
|
|
/* only set as master if we need to resample */
|
|
GST_DEBUG_OBJECT (sink, "Setting clock as master");
|
|
gst_clock_set_master (sink->provided_clock, clock);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* start ringbuffer so we can start slaving right away when we need to */
|
|
gst_ring_buffer_start (sink->ringbuffer);
|
|
|
|
done:
|
|
return GST_STATE_CHANGE_SUCCESS;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_audio_sink_do_play (GstBaseAudioSink * sink)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
ret = gst_base_audio_sink_async_play (GST_BASE_SINK_CAST (sink));
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_audio_sink_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (sink->ringbuffer == NULL) {
|
|
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
|
|
}
|
|
if (!gst_ring_buffer_open_device (sink->ringbuffer))
|
|
goto open_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
sink->next_sample = -1;
|
|
sink->priv->last_align = -1;
|
|
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
|
|
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
gst_base_audio_sink_do_play (sink);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* need to take the lock so we don't interfere with an
|
|
* async play */
|
|
GST_OBJECT_LOCK (sink);
|
|
/* ringbuffer cannot start anymore */
|
|
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
|
|
gst_ring_buffer_pause (sink->ringbuffer);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* make sure we unblock before calling the parent state change
|
|
* so it can grab the STREAM_LOCK */
|
|
gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* stop slaving ourselves to the master, if any */
|
|
gst_clock_set_master (sink->provided_clock, NULL);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_ring_buffer_release (sink->ringbuffer);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
/* we release again here because the aqcuire happens when setting the
|
|
* caps, which happens before we commit the state to PAUSED and thus the
|
|
* PAUSED->READY state change (see above, where we release the ringbuffer)
|
|
* might not be called when we get here. */
|
|
gst_ring_buffer_release (sink->ringbuffer);
|
|
gst_ring_buffer_close_device (sink->ringbuffer);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
open_failed:
|
|
{
|
|
/* subclass must post a meaningfull error message */
|
|
GST_DEBUG_OBJECT (sink, "open failed");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|