mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-15 04:46:32 +00:00
f04f86f3ee
This reverts commit 8e923a8e2d.
This caused regressions, see #3303.
Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6356>
2526 lines
81 KiB
C
2526 lines
81 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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* gstaudiobasesink.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudiobasesink
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* @title: GstAudioBaseSink
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* @short_description: Base class for audio sinks
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* @see_also: #GstAudioSink, #GstAudioRingBuffer.
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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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#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 <gst/audio/audio.h>
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#include "gstaudiobasesink.h"
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GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
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#define GST_CAT_DEFAULT gst_audio_base_sink_debug
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struct _GstAudioBaseSinkPrivate
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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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GstAudioBaseSinkSlaveMethod 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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gboolean sync_latency;
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GstClockTime eos_time;
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/* number of microseconds we allow clock slaving to drift
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* before resyncing */
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guint64 drift_tolerance;
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/* number of nanoseconds we allow timestamps to drift
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* before resyncing */
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GstClockTime alignment_threshold;
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/* time of the previous detected discont candidate */
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GstClockTime discont_time;
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/* number of nanoseconds to wait until creating a discontinuity */
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GstClockTime discont_wait;
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/* custom slaving algorithm callback */
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GstAudioBaseSinkCustomSlavingCallback custom_slaving_callback;
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gpointer custom_slaving_cb_data;
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GDestroyNotify custom_slaving_cb_notify;
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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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/* FIXME: 2.0, 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_AUDIO_BASE_SINK_SLAVE_SKEW
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/* FIXME, enable pull mode when clock slaving and trick modes are figured out */
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#define DEFAULT_CAN_ACTIVATE_PULL FALSE
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/* when timestamps drift for more than 40ms we resync. This should
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* be enough to compensate for timestamp rounding errors. */
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#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
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/* when clock slaving drift for more than 40ms we resync. This is
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* a reasonable default */
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#define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND)
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/* allow for one second before resyncing to see if the timestamps drift will
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* fix itself, or is a permanent offset */
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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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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PROP_CAN_ACTIVATE_PULL,
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PROP_ALIGNMENT_THRESHOLD,
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PROP_DRIFT_TOLERANCE,
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PROP_DISCONT_WAIT,
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PROP_LAST
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};
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#define _do_init \
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "audiobasesink", 0, "audiobasesink element");
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#define gst_audio_base_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink,
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GST_TYPE_BASE_SINK, G_ADD_PRIVATE (GstAudioBaseSink) _do_init);
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static void gst_audio_base_sink_dispose (GObject * object);
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static void gst_audio_base_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_base_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink,
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gboolean active);
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static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
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query);
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static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem);
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static inline void gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink);
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static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
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GstAudioBaseSink * sink);
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static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
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guint8 * data, guint len, gpointer user_data);
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static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink,
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GstBuffer * buffer);
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static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink,
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GstBuffer * buffer);
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static gboolean gst_audio_base_sink_event (GstBaseSink * bsink,
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GstEvent * event);
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static GstFlowReturn gst_audio_base_sink_wait_event (GstBaseSink * bsink,
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GstEvent * event);
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static void gst_audio_base_sink_get_times (GstBaseSink * bsink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink,
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GstCaps * caps);
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static GstCaps *gst_audio_base_sink_fixate (GstBaseSink * bsink,
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GstCaps * caps);
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static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
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GstQuery * query);
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/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
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static void
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gst_audio_base_sink_class_init (GstAudioBaseSinkClass * 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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gobject_class->set_property = gst_audio_base_sink_set_property;
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gobject_class->get_property = gst_audio_base_sink_get_property;
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gobject_class->dispose = gst_audio_base_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, this is the minimum "
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"latency that the sink reports", 1, 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_LATENCY_TIME,
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g_param_spec_int64 ("latency-time", "Latency Time",
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"The minimum amount of data to write in each iteration "
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"in microseconds", 1, G_MAXINT64, DEFAULT_LATENCY_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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 | G_PARAM_STATIC_STRINGS));
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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 used to match the rate of the masterclock",
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GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
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g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
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"Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAudioBaseSink:drift-tolerance:
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*
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* Controls the amount of time in microseconds that clocks are allowed
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* to drift before resynchronisation happens.
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*/
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g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
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g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
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"Tolerance for clock drift in microseconds", 1,
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G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAudioBaseSink:alignment_threshold:
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*
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* Controls the amount of time in nanoseconds that timestamps are allowed
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* to drift from their ideal time before choosing not to align them.
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*/
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g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
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g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
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"Timestamp alignment threshold in nanoseconds", 1,
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G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAudioBaseSink:discont-wait:
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*
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* A window of time in nanoseconds to wait before creating a discontinuity as
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* a result of breaching the drift-tolerance.
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*/
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g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
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g_param_spec_uint64 ("discont-wait", "Discont Wait",
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"Window of time in nanoseconds to wait before "
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"creating a discontinuity", 0,
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G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state);
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gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock);
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gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query);
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gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_sink_fixate);
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gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_sink_setcaps);
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gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_sink_event);
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gstbasesink_class->wait_event =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_wait_event);
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gstbasesink_class->get_times =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_get_times);
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gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_audio_base_sink_preroll);
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gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_audio_base_sink_render);
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gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query_pad);
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gstbasesink_class->activate_pull =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_activate_pull);
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/* ref class from a thread-safe context to work around missing bit of
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* thread-safety in GObject */
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g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
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g_type_class_ref (GST_TYPE_AUDIO_RING_BUFFER);
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}
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static void
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gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
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{
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GstBaseSink *basesink = GST_BASE_SINK_CAST (audiobasesink);
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audiobasesink->priv =
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gst_audio_base_sink_get_instance_private (audiobasesink);
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audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
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audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
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audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
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audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
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audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
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audiobasesink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
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audiobasesink->priv->custom_slaving_callback = NULL;
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audiobasesink->priv->custom_slaving_cb_data = NULL;
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audiobasesink->priv->custom_slaving_cb_notify = NULL;
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audiobasesink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
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(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, audiobasesink,
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NULL);
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basesink->can_activate_push = TRUE;
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basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
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gst_base_sink_set_last_sample_enabled (basesink, FALSE);
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if (DEFAULT_PROVIDE_CLOCK)
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GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
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else
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GST_OBJECT_FLAG_UNSET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
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}
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static void
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gst_audio_base_sink_dispose (GObject * object)
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{
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GstAudioBaseSink *sink;
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sink = GST_AUDIO_BASE_SINK (object);
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if (sink->priv->custom_slaving_cb_notify)
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sink->priv->custom_slaving_cb_notify (sink->priv->custom_slaving_cb_data);
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if (sink->provided_clock) {
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gst_audio_clock_invalidate (GST_AUDIO_CLOCK (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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}
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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_audio_base_sink_provide_clock (GstElement * elem)
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{
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GstAudioBaseSink *sink;
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GstClock *clock;
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sink = GST_AUDIO_BASE_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_audio_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 (!GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_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_audio_base_sink_is_self_provided_clock (GstAudioBaseSink * sink)
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{
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return (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
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GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
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(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time);
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}
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static gboolean
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gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
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{
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gboolean res = FALSE;
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GstAudioBaseSink *basesink;
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basesink = GST_AUDIO_BASE_SINK (bsink);
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CONVERT:
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{
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
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GST_LOG_OBJECT (basesink, "query convert");
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if (basesink->ringbuffer) {
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
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res =
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gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
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src_val, dest_fmt, &dest_val);
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if (res) {
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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}
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}
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break;
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}
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default:
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res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
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break;
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}
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return res;
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}
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static gboolean
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gst_audio_base_sink_query (GstElement * element, GstQuery * query)
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{
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gboolean res = FALSE;
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GstAudioBaseSink *basesink;
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basesink = GST_AUDIO_BASE_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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/* 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 base_latency, min_latency, max_latency;
|
|
|
|
/* we and upstream are both live, adjust the min_latency */
|
|
if (live && us_live) {
|
|
GstAudioRingBufferSpec *spec;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
if (!basesink->ringbuffer || !basesink->ringbuffer->spec.info.rate) {
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"we are not negotiated, can't report latency yet");
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
spec = &basesink->ringbuffer->spec;
|
|
|
|
basesink->priv->us_latency = min_l;
|
|
|
|
base_latency =
|
|
gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
|
|
GST_SECOND, spec->info.rate * spec->info.bpf);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
/* we cannot go lower than the buffer size and the min peer latency */
|
|
min_latency = base_latency + min_l;
|
|
/* the max latency is the max of the peer, we can delay an infinite
|
|
* amount of time. */
|
|
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
|
|
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"peer min %" GST_TIME_FORMAT ", our min latency: %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
|
|
GST_TIME_ARGS (min_latency));
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"peer max %" GST_TIME_FORMAT ", our max latency: %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
|
|
GST_TIME_ARGS (max_latency));
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"peer or we are not live, don't care about latency");
|
|
min_latency = min_l;
|
|
max_latency = max_l;
|
|
}
|
|
gst_query_set_latency (query, live, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
GST_LOG_OBJECT (basesink, "query convert");
|
|
|
|
if (basesink->ringbuffer) {
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
|
|
res =
|
|
gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
|
|
src_val, dest_fmt, &dest_val);
|
|
if (res) {
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
|
|
break;
|
|
}
|
|
|
|
done:
|
|
return res;
|
|
}
|
|
|
|
|
|
/* we call this function without holding the lock on sink for performance
|
|
* reasons. Try hard to not deal with and invalid ringbuffer and rate. */
|
|
static GstClockTime
|
|
gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
|
|
{
|
|
guint64 raw, samples;
|
|
guint delay;
|
|
GstClockTime result;
|
|
GstAudioRingBuffer *ringbuffer;
|
|
gint rate;
|
|
|
|
if ((ringbuffer = sink->ringbuffer) == NULL)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
if ((rate = ringbuffer->spec.info.rate) == 0)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
/* our processed samples are always increasing */
|
|
raw = samples = gst_audio_ring_buffer_samples_done (ringbuffer);
|
|
|
|
/* the number of samples not yet processed, this is still queued in the
|
|
* device (not played for playback). */
|
|
delay = gst_audio_ring_buffer_delay (ringbuffer);
|
|
|
|
if (G_LIKELY (samples >= delay))
|
|
samples -= delay;
|
|
else
|
|
samples = 0;
|
|
|
|
result = gst_util_uint64_scale_int (samples, GST_SECOND, rate);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
|
|
G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
|
|
raw, delay, samples, GST_TIME_ARGS (result));
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_provide_clock:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @provide: new state
|
|
*
|
|
* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
|
|
* gst_element_provide_clock() will return a clock that reflects the datarate
|
|
* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return
|
|
* NULL.
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
|
|
gboolean provide)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if (provide)
|
|
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
|
|
else
|
|
GST_OBJECT_FLAG_UNSET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_get_provide_clock:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Queries whether @sink will provide a clock or not. See also
|
|
* gst_audio_base_sink_set_provide_clock.
|
|
*
|
|
* Returns: %TRUE if @sink will provide a clock.
|
|
*/
|
|
gboolean
|
|
gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_slave_method:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @method: the new slave method
|
|
*
|
|
* Controls how clock slaving will be performed in @sink.
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
|
|
GstAudioBaseSinkSlaveMethod method)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->slave_method = method;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_get_slave_method:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Get the current slave method used by @sink.
|
|
*
|
|
* Returns: The current slave method used by @sink.
|
|
*/
|
|
GstAudioBaseSinkSlaveMethod
|
|
gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
|
|
{
|
|
GstAudioBaseSinkSlaveMethod result;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = sink->priv->slave_method;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_drift_tolerance:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @drift_tolerance: the new drift tolerance in microseconds
|
|
*
|
|
* Controls the sink's drift tolerance.
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
|
|
gint64 drift_tolerance)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->drift_tolerance = drift_tolerance;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_get_drift_tolerance:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Get the current drift tolerance, in microseconds, used by @sink.
|
|
*
|
|
* Returns: The current drift tolerance used by @sink.
|
|
*/
|
|
gint64
|
|
gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
|
|
{
|
|
gint64 result;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = sink->priv->drift_tolerance;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_alignment_threshold:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @alignment_threshold: the new alignment threshold in nanoseconds
|
|
*
|
|
* Controls the sink's alignment threshold.
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
|
|
GstClockTime alignment_threshold)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (alignment_threshold));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->alignment_threshold = alignment_threshold;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_get_alignment_threshold:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Get the current alignment threshold, in nanoseconds, used by @sink.
|
|
*
|
|
* Returns: The current alignment threshold used by @sink.
|
|
*/
|
|
GstClockTime
|
|
gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
|
|
{
|
|
GstClockTime result;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), GST_CLOCK_TIME_NONE);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = sink->priv->alignment_threshold;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_discont_wait:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @discont_wait: the new discont wait in nanoseconds
|
|
*
|
|
* Controls how long the sink will wait before creating a discontinuity.
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
|
|
GstClockTime discont_wait)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (discont_wait));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->discont_wait = discont_wait;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_set_custom_slaving_callback:
|
|
* @sink: a #GstAudioBaseSink
|
|
* @callback: a #GstAudioBaseSinkCustomSlavingCallback
|
|
* @user_data: user data passed to the callback
|
|
* @notify : called when user_data becomes unused
|
|
*
|
|
* Sets the custom slaving callback. This callback will
|
|
* be invoked if the slave-method property is set to
|
|
* GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink
|
|
* receives and plays samples.
|
|
*
|
|
* Setting the callback to NULL causes the sink to
|
|
* behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE
|
|
* method were used.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
void
|
|
gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink,
|
|
GstAudioBaseSinkCustomSlavingCallback callback,
|
|
gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->custom_slaving_callback = callback;
|
|
sink->priv->custom_slaving_cb_data = user_data;
|
|
sink->priv->custom_slaving_cb_notify = notify;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_custom_cb_report_discont (GstAudioBaseSink * sink,
|
|
GstAudioBaseSinkDiscontReason discont_reason)
|
|
{
|
|
if ((sink->priv->custom_slaving_callback != NULL) &&
|
|
(sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_CUSTOM)) {
|
|
sink->priv->custom_slaving_callback (sink, GST_CLOCK_TIME_NONE,
|
|
GST_CLOCK_TIME_NONE, NULL, discont_reason,
|
|
sink->priv->custom_slaving_cb_data);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_report_device_failure:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Informs this base class that the audio output device has failed for
|
|
* some reason, causing a discontinuity (for example, because the device
|
|
* recovered from the error, but lost all contents of its ring buffer).
|
|
* This function is typically called by derived classes, and is useful
|
|
* for the custom slave method.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
void
|
|
gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
gst_audio_base_sink_custom_cb_report_discont (sink,
|
|
GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_get_discont_wait:
|
|
* @sink: a #GstAudioBaseSink
|
|
*
|
|
* Get the current discont wait, in nanoseconds, used by @sink.
|
|
*
|
|
* Returns: The current discont wait used by @sink.
|
|
*/
|
|
GstClockTime
|
|
gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
|
|
{
|
|
GstClockTime result;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
result = sink->priv->discont_wait;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioBaseSink *sink;
|
|
|
|
sink = GST_AUDIO_BASE_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_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_SLAVE_METHOD:
|
|
gst_audio_base_sink_set_slave_method (sink, g_value_get_enum (value));
|
|
break;
|
|
case PROP_CAN_ACTIVATE_PULL:
|
|
GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_DRIFT_TOLERANCE:
|
|
gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
|
|
break;
|
|
case PROP_ALIGNMENT_THRESHOLD:
|
|
gst_audio_base_sink_set_alignment_threshold (sink,
|
|
g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_DISCONT_WAIT:
|
|
gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioBaseSink *sink;
|
|
|
|
sink = GST_AUDIO_BASE_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_audio_base_sink_get_provide_clock (sink));
|
|
break;
|
|
case PROP_SLAVE_METHOD:
|
|
g_value_set_enum (value, gst_audio_base_sink_get_slave_method (sink));
|
|
break;
|
|
case PROP_CAN_ACTIVATE_PULL:
|
|
g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
|
|
break;
|
|
case PROP_DRIFT_TOLERANCE:
|
|
g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink));
|
|
break;
|
|
case PROP_ALIGNMENT_THRESHOLD:
|
|
g_value_set_uint64 (value,
|
|
gst_audio_base_sink_get_alignment_threshold (sink));
|
|
break;
|
|
case PROP_DISCONT_WAIT:
|
|
g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
|
|
GstAudioRingBufferSpec *spec;
|
|
GstClockTime now, internal_time;
|
|
GstClockTime crate_num, crate_denom;
|
|
|
|
if (!sink->ringbuffer)
|
|
return FALSE;
|
|
|
|
spec = &sink->ringbuffer->spec;
|
|
|
|
if (G_UNLIKELY (spec->caps && gst_caps_is_equal (spec->caps, caps))) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"Ringbuffer caps haven't changed, skipping reconfiguration");
|
|
return TRUE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
|
|
|
|
/* get current time, updates the last_time. When the subclass has a clock that
|
|
* restarts from 0 when a new format is negotiated, it will call
|
|
* gst_audio_clock_reset() which will use this last_time to create an offset
|
|
* so that time from the clock keeps on increasing monotonically. */
|
|
now = gst_clock_get_time (sink->provided_clock);
|
|
internal_time = gst_clock_get_internal_time (sink->provided_clock);
|
|
|
|
GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
|
|
|
|
/* release old ringbuffer */
|
|
gst_audio_ring_buffer_pause (sink->ringbuffer);
|
|
gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
|
|
gst_audio_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_audio_ring_buffer_parse_caps (spec, caps))
|
|
goto parse_error;
|
|
|
|
gst_audio_ring_buffer_debug_spec_buff (spec);
|
|
|
|
GST_DEBUG_OBJECT (sink, "acquire ringbuffer");
|
|
if (!gst_audio_ring_buffer_acquire (sink->ringbuffer, spec))
|
|
goto acquire_error;
|
|
|
|
/* If we use our own clock, we need to adjust the offset since it will now
|
|
* restart from zero */
|
|
if (gst_audio_base_sink_is_self_provided_clock (sink))
|
|
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
|
|
|
|
/* We need to resync since the ringbuffer restarted */
|
|
gst_audio_base_sink_reset_sync (sink);
|
|
|
|
gst_audio_base_sink_custom_cb_report_discont (sink,
|
|
GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS);
|
|
|
|
if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
|
|
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
|
|
gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
|
|
}
|
|
|
|
/* due to possible changes in the spec file we should recalibrate the clock */
|
|
gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
|
|
&crate_num, &crate_denom);
|
|
gst_clock_set_calibration (sink->provided_clock,
|
|
internal_time, now, crate_num, crate_denom);
|
|
|
|
/* calculate actual latency and buffer times.
|
|
* FIXME: In 2.0, store the latency_time internally in ns */
|
|
spec->latency_time = gst_util_uint64_scale (spec->segsize,
|
|
(GST_SECOND / GST_USECOND), spec->info.rate * spec->info.bpf);
|
|
|
|
spec->buffer_time = spec->segtotal * spec->latency_time;
|
|
|
|
gst_audio_ring_buffer_debug_spec_buff (spec);
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (bsink),
|
|
gst_message_new_latency (GST_OBJECT (bsink)));
|
|
|
|
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 GstCaps *
|
|
gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstStructure *s;
|
|
gint width, depth;
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
|
|
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);
|
|
|
|
caps = GST_BASE_SINK_CLASS (parent_class)->fixate (bsink, caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static inline void
|
|
gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink)
|
|
{
|
|
sink->next_sample = -1;
|
|
sink->priv->eos_time = -1;
|
|
sink->priv->discont_time = -1;
|
|
sink->priv->avg_skew = -1;
|
|
sink->priv->last_align = 0;
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
/* our clock sync is a bit too much for the base class to handle so
|
|
* we implement it ourselves. */
|
|
*start = GST_CLOCK_TIME_NONE;
|
|
*end = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_force_start (GstAudioBaseSink * sink)
|
|
{
|
|
/* Set the eos_rendering flag so sub-classes definitely start the clock.
|
|
* FIXME 2.0: Pass this as a flag to gst_audio_ring_buffer_start() */
|
|
g_atomic_int_set (&sink->eos_rendering, 1);
|
|
gst_audio_ring_buffer_start (sink->ringbuffer);
|
|
g_atomic_int_set (&sink->eos_rendering, 0);
|
|
}
|
|
|
|
/* This waits for the drain to happen and can be canceled */
|
|
static GstFlowReturn
|
|
gst_audio_base_sink_drain (GstAudioBaseSink * sink)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
if (!sink->ringbuffer)
|
|
return ret;
|
|
if (!sink->ringbuffer->spec.info.rate)
|
|
return ret;
|
|
|
|
/* if PLAYING is interrupted,
|
|
* arrange to have clock running when going to PLAYING again */
|
|
g_atomic_int_set (&sink->eos_rendering, 1);
|
|
|
|
/* need to start playback before we can drain, but only when
|
|
* we have successfully negotiated a format and thus acquired the
|
|
* ringbuffer. */
|
|
if (gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
|
|
gst_audio_ring_buffer_start (sink->ringbuffer);
|
|
|
|
if (sink->priv->eos_time != -1) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last sample time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (sink->priv->eos_time));
|
|
|
|
/* wait for the EOS time to be reached, this is the time when the last
|
|
* sample is played. */
|
|
ret = gst_base_sink_wait (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
|
|
|
|
GST_DEBUG_OBJECT (sink, "drained audio");
|
|
}
|
|
g_atomic_int_set (&sink->eos_rendering, 0);
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
|
|
{
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gboolean clear_force_start_flag = FALSE;
|
|
|
|
/* For both gap and EOS events, make sure the ringbuffer is running
|
|
* before trying to wait on the event! */
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
case GST_EVENT_GAP:
|
|
/* We must have a negotiated format before starting the ringbuffer */
|
|
if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))) {
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL),
|
|
("Sink not negotiated before %s event.",
|
|
GST_EVENT_TYPE_NAME (event)));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
gst_audio_base_sink_force_start (sink);
|
|
/* Make sure the ringbuffer will start again if interrupted during event_wait() */
|
|
g_atomic_int_set (&sink->eos_rendering, 1);
|
|
clear_force_start_flag = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_BASE_SINK_CLASS (parent_class)->wait_event (bsink, event);
|
|
if (ret != GST_FLOW_OK)
|
|
goto done;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
/* now wait till we played everything */
|
|
ret = gst_audio_base_sink_drain (sink);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
done:
|
|
if (clear_force_start_flag)
|
|
g_atomic_int_set (&sink->eos_rendering, 0);
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
|
|
{
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
if (sink->ringbuffer)
|
|
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* always resync on sample after a flush */
|
|
gst_audio_base_sink_reset_sync (sink);
|
|
|
|
gst_audio_base_sink_custom_cb_report_discont (sink,
|
|
GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH);
|
|
|
|
if (sink->ringbuffer)
|
|
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
|
|
{
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
|
|
|
|
if (!gst_audio_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_audio_base_sink_get_offset (GstAudioBaseSink * sink)
|
|
{
|
|
guint64 sample, sps;
|
|
gint writeseg, segdone;
|
|
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)
|
|
{
|
|
/* 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;
|
|
}
|
|
return external;
|
|
}
|
|
|
|
|
|
/* apply the clock offset and invoke a custom callback
|
|
* which might also request changes to the playout pointer
|
|
*
|
|
* this reuses code from the skewing algorithm, but leaves
|
|
* decision on whether or not to skew (and how much to skew)
|
|
* up to the callback */
|
|
static void
|
|
gst_audio_base_sink_custom_slaving (GstAudioBaseSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
GstClockTime cinternal, cexternal, crate_num, crate_denom;
|
|
GstClockTime etime, itime;
|
|
GstClockTimeDiff requested_skew;
|
|
gint driftsamples;
|
|
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_audio_clock_get_time (GST_AUDIO_CLOCK (sink->provided_clock));
|
|
itime =
|
|
gst_audio_clock_adjust (GST_AUDIO_CLOCK (sink->provided_clock), itime);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
|
|
" cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
|
|
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
|
|
|
|
/* make sure we never go below 0 */
|
|
etime = etime > cexternal ? etime - cexternal : 0;
|
|
itime = itime > cinternal ? itime - cinternal : 0;
|
|
|
|
/* don't do any skewing unless the callback explicitly requests one */
|
|
requested_skew = 0;
|
|
|
|
if (sink->priv->custom_slaving_callback != NULL) {
|
|
sink->priv->custom_slaving_callback (sink, etime, itime, &requested_skew,
|
|
FALSE, sink->priv->custom_slaving_cb_data);
|
|
GST_DEBUG_OBJECT (sink, "custom slaving requested skew %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (requested_skew));
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"no custom slaving callback set - clock drift will not be compensated");
|
|
}
|
|
|
|
if (requested_skew > 0) {
|
|
cexternal = (cexternal > requested_skew) ? (cexternal - requested_skew) : 0;
|
|
|
|
driftsamples =
|
|
(sink->ringbuffer->spec.info.rate * requested_skew) / GST_SECOND;
|
|
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 > driftsamples))
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
|
|
|
|
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
} else if (requested_skew < 0) {
|
|
cexternal += ABS (requested_skew);
|
|
|
|
driftsamples =
|
|
(sink->ringbuffer->spec.info.rate * ABS (requested_skew)) / GST_SECOND;
|
|
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 > driftsamples))
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, driftsamples, 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);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* algorithm to calculate sample positions that will result in resampling to
|
|
* match the clock rate of the master */
|
|
static void
|
|
gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
GstClockTime cinternal, cexternal;
|
|
GstClockTime crate_num, crate_denom;
|
|
|
|
/* FIXME, we can sample and add observations here or use the timeouts on the
|
|
* clock. No idea which one is better or more stable. The timeout seems more
|
|
* arbitrary but this one seems more demanding and does not work when there is
|
|
* no data coming in to the sink. */
|
|
#if 0
|
|
GstClockTime etime, itime;
|
|
gdouble r_squared;
|
|
|
|
/* sample clocks and figure out clock skew */
|
|
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
|
|
itime = gst_audio_clock_get_time (sink->provided_clock);
|
|
|
|
/* add new observation */
|
|
gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
|
|
#endif
|
|
|
|
/* 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);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
|
|
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_audio_base_sink_skew_slaving (GstAudioBaseSink * 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, drift, mdrift2;
|
|
gint driftsamples;
|
|
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_audio_clock_get_time (GST_AUDIO_CLOCK (sink->provided_clock));
|
|
itime =
|
|
gst_audio_clock_adjust (GST_AUDIO_CLOCK (sink->provided_clock), itime);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
|
|
" cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
|
|
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
|
|
|
|
/* make sure we never go below 0 */
|
|
etime = etime > cexternal ? etime - cexternal : 0;
|
|
itime = itime > cinternal ? itime - cinternal : 0;
|
|
|
|
/* do itime - etime.
|
|
* positive value means external clock goes slower
|
|
* negative value means external clock goes faster */
|
|
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 %" GST_STIME_FORMAT " avg %" GST_STIME_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), GST_STIME_ARGS (skew),
|
|
GST_STIME_ARGS (sink->priv->avg_skew));
|
|
|
|
/* the max drift we allow */
|
|
mdrift2 = (sink->priv->drift_tolerance * 1000) / 2;
|
|
|
|
/* adjust playout pointer based on skew */
|
|
if (sink->priv->avg_skew > mdrift2) {
|
|
/* master is running slower, move external time backwards */
|
|
GST_WARNING_OBJECT (sink,
|
|
"correct clock skew %" GST_STIME_FORMAT " > %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (sink->priv->avg_skew), GST_STIME_ARGS (mdrift2));
|
|
|
|
/* Move the external time backward by the average skew, but don't ever
|
|
* go negative. Moving the average skew by the same distance defines
|
|
* the new clock skew window center point. This allows the clock to
|
|
* drift equally into either direction after the correction. */
|
|
if (G_LIKELY (cexternal > sink->priv->avg_skew))
|
|
drift = sink->priv->avg_skew;
|
|
else
|
|
drift = cexternal;
|
|
cexternal -= drift;
|
|
sink->priv->avg_skew -= drift;
|
|
|
|
driftsamples = (sink->ringbuffer->spec.info.rate * drift) / GST_SECOND;
|
|
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 > driftsamples)
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
|
|
|
|
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
} else if (sink->priv->avg_skew < -mdrift2) {
|
|
/* master is running faster, move external time forwards */
|
|
GST_WARNING_OBJECT (sink,
|
|
"correct clock skew %" GST_STIME_FORMAT " < -%" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (sink->priv->avg_skew), GST_STIME_ARGS (mdrift2));
|
|
|
|
/* Move the external time forward by the average skew, and move the
|
|
* average skew by the same distance (which equals a reset to 0). This
|
|
* defines the new clock skew window center point. This allows the
|
|
* clock to drift equally into either direction after the correction. */
|
|
drift = -sink->priv->avg_skew;
|
|
cexternal += drift;
|
|
sink->priv->avg_skew = 0;
|
|
|
|
driftsamples = (sink->ringbuffer->spec.info.rate * drift) / GST_SECOND;
|
|
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 > driftsamples)
|
|
sink->next_sample = -1;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
|
|
G_GUINT64_FORMAT, last_align, driftsamples, 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);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* apply the clock offset but do no slaving otherwise */
|
|
static void
|
|
gst_audio_base_sink_none_slaving (GstAudioBaseSink * 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);
|
|
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
|
|
crate_num, crate_denom);
|
|
|
|
*srender_start = render_start;
|
|
*srender_stop = render_stop;
|
|
}
|
|
|
|
/* converts render_start and render_stop to their slaved values */
|
|
static void
|
|
gst_audio_base_sink_handle_slaving (GstAudioBaseSink * sink,
|
|
GstClockTime render_start, GstClockTime render_stop,
|
|
GstClockTime * srender_start, GstClockTime * srender_stop)
|
|
{
|
|
switch (sink->priv->slave_method) {
|
|
case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
|
|
gst_audio_base_sink_resample_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
|
|
gst_audio_base_sink_skew_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
case GST_AUDIO_BASE_SINK_SLAVE_NONE:
|
|
gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
case GST_AUDIO_BASE_SINK_SLAVE_CUSTOM:
|
|
gst_audio_base_sink_custom_slaving (sink, render_start, render_stop,
|
|
srender_start, srender_stop);
|
|
break;
|
|
default:
|
|
g_warning ("unknown slaving method %d", sink->priv->slave_method);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* must be called with LOCK */
|
|
static GstFlowReturn
|
|
gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
|
|
{
|
|
GstClock *clock;
|
|
GstClockReturn status;
|
|
GstClockTime time, render_delay;
|
|
GstFlowReturn ret;
|
|
GstAudioBaseSink *sink;
|
|
GstClockTime itime, etime;
|
|
GstClockTime rate_num, rate_denom;
|
|
GstClockTimeDiff jitter;
|
|
|
|
sink = GST_AUDIO_BASE_SINK (bsink);
|
|
|
|
clock = GST_ELEMENT_CLOCK (sink);
|
|
if (G_UNLIKELY (clock == NULL))
|
|
goto no_clock;
|
|
|
|
/* we provided the global clock, don't need to do anything special */
|
|
if (clock == sink->provided_clock)
|
|
goto no_slaving;
|
|
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
do {
|
|
GST_DEBUG_OBJECT (sink, "checking preroll");
|
|
|
|
ret = gst_base_sink_do_preroll (bsink, obj);
|
|
if (ret != GST_FLOW_OK)
|
|
goto flushing;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
time = sink->priv->us_latency;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* Renderdelay is added onto our own latency, and needs
|
|
* to be subtracted as well */
|
|
render_delay = gst_base_sink_get_render_delay (bsink);
|
|
|
|
if (G_LIKELY (time > render_delay))
|
|
time -= render_delay;
|
|
else
|
|
time = 0;
|
|
|
|
/* preroll done, we can sync since we are in PLAYING now. */
|
|
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (time));
|
|
|
|
/* wait for the clock, this can be interrupted because we got shut down or
|
|
* we PAUSED. */
|
|
status = gst_base_sink_wait_clock (bsink, time, &jitter);
|
|
|
|
GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
|
|
GST_TIME_ARGS (jitter));
|
|
|
|
/* invalid time, no clock or sync disabled, just continue then */
|
|
if (status == GST_CLOCK_BADTIME)
|
|
break;
|
|
|
|
/* waiting could have been interrupted and we can be flushing now */
|
|
if (G_UNLIKELY (bsink->flushing))
|
|
goto flushing;
|
|
|
|
/* retry if we got unscheduled, which means we did not reach the timeout
|
|
* yet. if some other error occurs, we continue. */
|
|
} while (status == GST_CLOCK_UNSCHEDULED);
|
|
|
|
GST_DEBUG_OBJECT (sink, "latency synced");
|
|
|
|
/* We might need to take the object lock within gst_audio_clock_get_time(),
|
|
* so call that before we take it again */
|
|
itime = gst_audio_clock_get_time (GST_AUDIO_CLOCK (sink->provided_clock));
|
|
itime =
|
|
gst_audio_clock_adjust (GST_AUDIO_CLOCK (sink->provided_clock), itime);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
|
|
/* when we prerolled in time, we can accurately set the calibration,
|
|
* our internal clock should exactly have been the latency (== the running
|
|
* time of the external clock) */
|
|
etime = GST_ELEMENT_CAST (sink)->base_time + time;
|
|
|
|
if (status == GST_CLOCK_EARLY) {
|
|
/* when we prerolled late, we have to take into account the lateness */
|
|
GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
|
|
etime += jitter;
|
|
}
|
|
|
|
/* start ringbuffer so we can start slaving right away when we need to */
|
|
gst_audio_base_sink_force_start (sink);
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
|
|
|
|
/* copy the original calibrated rate but update the internal and external
|
|
* times. */
|
|
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_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
|
|
/* only set as master when we are resampling */
|
|
GST_DEBUG_OBJECT (sink, "Setting clock as master");
|
|
gst_clock_set_master (sink->provided_clock, clock);
|
|
break;
|
|
case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
|
|
case GST_AUDIO_BASE_SINK_SLAVE_NONE:
|
|
case GST_AUDIO_BASE_SINK_SLAVE_CUSTOM:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
gst_audio_base_sink_reset_sync (sink);
|
|
|
|
gst_audio_base_sink_custom_cb_report_discont (sink,
|
|
GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY);
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we have no clock");
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_slaving:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we are not slaved");
|
|
return GST_FLOW_OK;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we are flushing");
|
|
GST_OBJECT_LOCK (sink);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
static gint64
|
|
gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
|
|
GstClockTime sample_offset)
|
|
{
|
|
GstAudioRingBuffer *ringbuf = sink->ringbuffer;
|
|
gint64 align;
|
|
gint64 sample_diff;
|
|
gint64 max_sample_diff;
|
|
gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
|
|
gint64 samples_done = segdone * (gint64) ringbuf->samples_per_seg;
|
|
gint64 headroom = sample_offset - samples_done;
|
|
gboolean allow_align = TRUE;
|
|
gboolean discont = FALSE;
|
|
gint rate;
|
|
|
|
/* now try to align the sample to the previous one. */
|
|
|
|
/* calc align with previous sample and determine how big the
|
|
* difference is. */
|
|
align = sink->next_sample - sample_offset;
|
|
sample_diff = ABS (align);
|
|
|
|
/* calculate the max allowed drift in units of samples. */
|
|
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
|
|
max_sample_diff = gst_util_uint64_scale_int (sink->priv->alignment_threshold,
|
|
rate, GST_SECOND);
|
|
|
|
/* don't align if it means writing behind the read-segment */
|
|
if (sample_diff > headroom && align < 0)
|
|
allow_align = FALSE;
|
|
|
|
if (G_UNLIKELY (sample_diff >= max_sample_diff)) {
|
|
/* wait before deciding to make a discontinuity */
|
|
if (sink->priv->discont_wait > 0) {
|
|
GstClockTime time = gst_util_uint64_scale_int (sample_offset,
|
|
GST_SECOND, rate);
|
|
if (sink->priv->discont_time == -1) {
|
|
/* discont candidate */
|
|
sink->priv->discont_time = time;
|
|
} else if (time - sink->priv->discont_time >= sink->priv->discont_wait) {
|
|
/* discont_wait expired, discontinuity detected */
|
|
discont = TRUE;
|
|
sink->priv->discont_time = -1;
|
|
}
|
|
} else {
|
|
discont = TRUE;
|
|
}
|
|
} else if (G_UNLIKELY (sink->priv->discont_time != -1)) {
|
|
/* we have had a discont, but are now back on track! */
|
|
sink->priv->discont_time = -1;
|
|
}
|
|
|
|
if (G_LIKELY (!discont && allow_align)) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
|
|
G_GINT64_FORMAT, align, max_sample_diff);
|
|
} else {
|
|
gint64 diff_s G_GNUC_UNUSED;
|
|
|
|
/* calculate sample diff in seconds for error message */
|
|
diff_s = gst_util_uint64_scale_int (sample_diff, GST_SECOND, rate);
|
|
|
|
/* timestamps drifted apart from previous samples too much, we need to
|
|
* resync. We log this as an element warning. */
|
|
GST_WARNING_OBJECT (sink,
|
|
"Unexpected discontinuity in audio timestamps of "
|
|
"%s%" GST_TIME_FORMAT ", resyncing",
|
|
sample_offset > sink->next_sample ? "+" : "-", GST_TIME_ARGS (diff_s));
|
|
align = 0;
|
|
|
|
gst_audio_base_sink_custom_cb_report_discont (sink,
|
|
GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT);
|
|
}
|
|
|
|
return align;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|
{
|
|
GstClockTime time, stop, render_start, render_stop, sample_offset;
|
|
GstClockTimeDiff sync_offset, ts_offset;
|
|
GstAudioBaseSinkClass *bclass;
|
|
GstAudioBaseSink *sink;
|
|
GstAudioRingBuffer *ringbuf;
|
|
gint64 diff, align;
|
|
guint64 ctime, cstop;
|
|
gsize offset;
|
|
GstMapInfo info;
|
|
gsize size;
|
|
guint samples, written;
|
|
gint bpf, rate;
|
|
gint accum;
|
|
gint out_samples;
|
|
GstClockTime base_time, render_delay, latency;
|
|
GstClock *clock;
|
|
gboolean sync, slaved, align_next;
|
|
GstFlowReturn ret;
|
|
GstSegment clip_seg;
|
|
gint64 time_offset;
|
|
GstBuffer *out = NULL;
|
|
|
|
sink = GST_AUDIO_BASE_SINK (bsink);
|
|
bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
|
|
|
|
ringbuf = sink->ringbuffer;
|
|
|
|
/* can't do anything when we don't have the device */
|
|
if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (ringbuf)))
|
|
goto wrong_state;
|
|
|
|
/* Wait for upstream latency before starting the ringbuffer, we do this so
|
|
* that we can align the first sample of the ringbuffer to the base_time +
|
|
* latency. */
|
|
GST_OBJECT_LOCK (sink);
|
|
base_time = GST_ELEMENT_CAST (sink)->base_time;
|
|
if (G_UNLIKELY (sink->priv->sync_latency)) {
|
|
ret = gst_audio_base_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
|
|
GST_OBJECT_UNLOCK (sink);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto sync_latency_failed;
|
|
/* only do this once until we are set back to PLAYING */
|
|
sink->priv->sync_latency = FALSE;
|
|
} else {
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/* Before we go on, let's see if we need to payload the data. If yes, we also
|
|
* need to unref the output buffer before leaving. */
|
|
if (bclass->payload) {
|
|
out = bclass->payload (sink, buf);
|
|
|
|
if (!out)
|
|
goto payload_failed;
|
|
|
|
buf = out;
|
|
}
|
|
|
|
bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
|
|
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
|
|
|
|
size = gst_buffer_get_size (buf);
|
|
if (G_UNLIKELY (size % bpf) != 0)
|
|
goto wrong_size;
|
|
|
|
samples = size / bpf;
|
|
|
|
time = GST_BUFFER_PTS (buf);
|
|
|
|
/* Last ditch attempt to ensure that we only play silence if
|
|
* we are in trickmode no-audio mode (or if a buffer is marked as a GAP)
|
|
* by dropping the buffer contents and rendering as a gap event instead */
|
|
if (G_UNLIKELY ((bsink->segment.flags & GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO)
|
|
|| (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))) {
|
|
GstClockTime duration;
|
|
GstEvent *event;
|
|
GstBaseSinkClass *bclass;
|
|
GST_DEBUG_OBJECT (bsink,
|
|
"Received GAP or ignoring audio for trickplay. Dropping contents");
|
|
|
|
duration = gst_util_uint64_scale_int (samples, GST_SECOND, rate);
|
|
event = gst_event_new_gap (time, duration);
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
ret = bclass->wait_event (bsink, event);
|
|
gst_event_unref (event);
|
|
|
|
/* Ensure we'll resync on the next buffer as if discont */
|
|
sink->next_sample = -1;
|
|
goto done;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"time %" GST_TIME_FORMAT ", start %"
|
|
GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time),
|
|
GST_TIME_ARGS (bsink->segment.start), samples);
|
|
|
|
offset = 0;
|
|
|
|
/* 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_audio_base_sink_get_offset (sink);
|
|
render_stop = render_start + samples;
|
|
GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
|
|
" Using render_start=%" G_GUINT64_FORMAT, size, render_start);
|
|
/* we don't have a start so we don't know stop either */
|
|
stop = -1;
|
|
goto no_align;
|
|
}
|
|
|
|
/* let's calc stop based on the number of samples in the buffer instead
|
|
* of trusting the DURATION */
|
|
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rate);
|
|
|
|
/* prepare the clipping segment. Since we will be subtracting ts-offset and
|
|
* device-delay later we scale the start and stop with those values so that we
|
|
* can correctly clip them */
|
|
clip_seg.format = GST_FORMAT_TIME;
|
|
clip_seg.start = bsink->segment.start;
|
|
clip_seg.stop = bsink->segment.stop;
|
|
clip_seg.duration = -1;
|
|
|
|
/* the sync offset is the combination of ts-offset and device-delay */
|
|
latency = gst_base_sink_get_latency (bsink);
|
|
ts_offset = gst_base_sink_get_ts_offset (bsink);
|
|
render_delay = gst_base_sink_get_render_delay (bsink);
|
|
sync_offset = ts_offset - render_delay + latency;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"sync-offset %" GST_STIME_FORMAT ", render-delay %" GST_TIME_FORMAT
|
|
", ts-offset %" GST_STIME_FORMAT, GST_STIME_ARGS (sync_offset),
|
|
GST_TIME_ARGS (render_delay), GST_STIME_ARGS (ts_offset));
|
|
|
|
/* compensate for ts-offset and device-delay when negative we need to
|
|
* clip. */
|
|
if (G_UNLIKELY (sync_offset < 0)) {
|
|
clip_seg.start += -sync_offset;
|
|
if (clip_seg.stop != -1)
|
|
clip_seg.stop += -sync_offset;
|
|
}
|
|
|
|
/* 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 */
|
|
if (G_UNLIKELY (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop,
|
|
&ctime, &cstop)))
|
|
goto out_of_segment;
|
|
|
|
/* see if some clipping happened */
|
|
diff = ctime - time;
|
|
if (G_UNLIKELY (diff > 0)) {
|
|
/* bring clipped time to samples */
|
|
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
|
|
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
|
|
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
|
|
samples -= diff;
|
|
offset += diff * bpf;
|
|
time = ctime;
|
|
}
|
|
diff = stop - cstop;
|
|
if (G_UNLIKELY (diff > 0)) {
|
|
/* bring clipped time to samples */
|
|
diff = gst_util_uint64_scale_int (diff, 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 (G_LIKELY ((clock = GST_ELEMENT_CLOCK (bsink))))
|
|
sync = bsink->sync;
|
|
else
|
|
sync = FALSE;
|
|
|
|
if (G_UNLIKELY (!sync)) {
|
|
/* no sync needed, play sample ASAP */
|
|
render_start = gst_audio_base_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_align;
|
|
}
|
|
|
|
/* 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);
|
|
|
|
if (render_start == GST_CLOCK_TIME_NONE || render_stop == GST_CLOCK_TIME_NONE)
|
|
goto too_late;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
|
|
|
/* store the time of the last sample, we'll use this to perform sync on the
|
|
* last sample when draining the buffer */
|
|
if (G_LIKELY (bsink->segment.rate >= 0.0)) {
|
|
sink->priv->eos_time = render_stop;
|
|
} else {
|
|
sink->priv->eos_time = render_start;
|
|
}
|
|
|
|
if (G_UNLIKELY (sync_offset != 0)) {
|
|
/* compensate for ts-offset and delay. We know this will not underflow
|
|
* because we clipped above. */
|
|
GST_DEBUG_OBJECT (sink,
|
|
"compensating for sync-offset %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (sync_offset));
|
|
render_start += sync_offset;
|
|
render_stop += sync_offset;
|
|
}
|
|
|
|
if (base_time != 0) {
|
|
GST_DEBUG_OBJECT (sink, "adding 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;
|
|
}
|
|
|
|
if (G_UNLIKELY ((slaved = (clock != sink->provided_clock)))) {
|
|
/* handle clock slaving */
|
|
gst_audio_base_sink_handle_slaving (sink, render_start, render_stop,
|
|
&render_start, &render_stop);
|
|
} else {
|
|
/* no slaving needed but we need to adapt to the clock calibration
|
|
* parameters */
|
|
gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
|
|
&render_start, &render_stop);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
|
|
|
/* bring to position in the ringbuffer */
|
|
time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
|
|
|
|
if (G_UNLIKELY (time_offset != 0)) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"apply time offset %" GST_STIME_FORMAT, GST_STIME_ARGS (time_offset));
|
|
|
|
if (render_start > time_offset)
|
|
render_start -= time_offset;
|
|
else
|
|
render_start = 0;
|
|
if (render_stop > time_offset)
|
|
render_stop -= time_offset;
|
|
else
|
|
render_stop = 0;
|
|
}
|
|
|
|
/* in some clock slaving cases, all late samples end up at 0 first,
|
|
* and subsequent ones align with that until threshold exceeded,
|
|
* and then sync back to 0 and so on, so avoid that altogether */
|
|
if (G_UNLIKELY (render_start == 0 && render_stop == 0))
|
|
goto too_late;
|
|
|
|
/* and bring the time to the rate corrected offset in the buffer */
|
|
render_start = gst_util_uint64_scale_int (render_start, rate, GST_SECOND);
|
|
render_stop = gst_util_uint64_scale_int (render_stop, rate, GST_SECOND);
|
|
|
|
/* If the slaving got us an interval spanning 0, render_start will
|
|
have been set to 0. So if render_start is 0, we check whether
|
|
render_stop is set to contain all samples. If not, we need to
|
|
drop samples to match. */
|
|
if (render_start == 0) {
|
|
guint nsamples = render_stop - render_start;
|
|
if (nsamples < samples) {
|
|
guint diff;
|
|
|
|
diff = samples - nsamples;
|
|
GST_DEBUG_OBJECT (bsink, "Clipped start: %u/%u samples", nsamples,
|
|
samples);
|
|
samples -= diff;
|
|
offset += diff * bpf;
|
|
}
|
|
}
|
|
|
|
/* 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 (G_LIKELY (bsink->segment.rate >= 0.0)) {
|
|
sample_offset = render_start;
|
|
if (G_LIKELY (bsink->segment.rate == 1.0))
|
|
render_stop = sample_offset + samples;
|
|
} else {
|
|
sample_offset = render_stop;
|
|
if (bsink->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_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_RESYNC))) {
|
|
GST_DEBUG_OBJECT (sink, "resync after discont/resync");
|
|
goto no_align;
|
|
}
|
|
|
|
/* resync when we don't know what to align the sample with */
|
|
if (G_UNLIKELY (sink->next_sample == -1)) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"no align possible: no previous sample position known");
|
|
goto no_align;
|
|
}
|
|
|
|
align = gst_audio_base_sink_get_alignment (sink, sample_offset);
|
|
sink->priv->last_align = align;
|
|
|
|
/* apply alignment */
|
|
render_start += align;
|
|
|
|
/* only align stop if we are not slaved to resample */
|
|
if (G_UNLIKELY (slaved
|
|
&& sink->priv->slave_method == GST_AUDIO_BASE_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;
|
|
|
|
/* we render the first or last sample first, depending on the rate */
|
|
if (G_LIKELY (bsink->segment.rate >= 0.0))
|
|
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;
|
|
gst_buffer_map (buf, &info, GST_MAP_READ);
|
|
do {
|
|
written =
|
|
gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
|
|
info.data + offset, samples, out_samples, &accum);
|
|
|
|
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
|
|
/* if we wrote all, we're done */
|
|
if (G_LIKELY (written == samples))
|
|
break;
|
|
|
|
/* else something interrupted us and we wait for preroll. */
|
|
if ((ret = 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;
|
|
|
|
/* update the output samples. FIXME, this will just skip them when pausing
|
|
* during trick mode */
|
|
if (out_samples > written) {
|
|
out_samples -= written;
|
|
accum = 0;
|
|
} else
|
|
break;
|
|
|
|
samples -= written;
|
|
offset += written * bpf;
|
|
} while (TRUE);
|
|
gst_buffer_unmap (buf, &info);
|
|
|
|
if (G_LIKELY (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 (G_UNLIKELY (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_audio_base_sink_force_start (sink);
|
|
}
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
done:
|
|
if (out)
|
|
gst_buffer_unref (out);
|
|
|
|
return ret;
|
|
|
|
/* 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));
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
too_late:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "dropping late sample");
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
/* ERRORS */
|
|
payload_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("failed to payload."));
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
wrong_state:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
|
|
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto done;
|
|
}
|
|
wrong_size:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "wrong size");
|
|
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
|
|
(NULL), ("sink received buffer of wrong size."));
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
gst_buffer_unmap (buf, &info);
|
|
goto done;
|
|
}
|
|
sync_latency_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "failed waiting for latency");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_base_sink_create_ringbuffer:
|
|
* @sink: a #GstAudioBaseSink.
|
|
*
|
|
* Create and return the #GstAudioRingBuffer 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: (transfer none) (nullable): The new ringbuffer of @sink.
|
|
*/
|
|
GstAudioRingBuffer *
|
|
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
|
|
{
|
|
GstAudioBaseSinkClass *bclass;
|
|
GstAudioRingBuffer *buffer = NULL;
|
|
|
|
bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
|
|
if (bclass->create_ringbuffer)
|
|
buffer = bclass->create_ringbuffer (sink);
|
|
|
|
if (buffer)
|
|
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
|
|
|
|
return buffer;
|
|
}
|
|
|
|
static void
|
|
gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
|
|
guint len, gpointer user_data)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstAudioBaseSink *sink;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn ret;
|
|
gsize size;
|
|
|
|
basesink = GST_BASE_SINK (user_data);
|
|
sink = GST_AUDIO_BASE_SINK (user_data);
|
|
|
|
GST_PAD_STREAM_LOCK (basesink->sinkpad);
|
|
|
|
/* 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 %u bytes offset %" G_GUINT64_FORMAT
|
|
" to fill audio buffer", len, basesink->offset);
|
|
ret =
|
|
gst_pad_pull_range (basesink->sinkpad, basesink->segment.position, len,
|
|
&buf);
|
|
|
|
if (ret != GST_FLOW_OK) {
|
|
if (ret == GST_FLOW_EOS)
|
|
goto eos;
|
|
else
|
|
goto error;
|
|
}
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
if (basesink->flushing)
|
|
goto flushing;
|
|
|
|
/* complete preroll and wait for PLAYING */
|
|
ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
|
|
if (ret != GST_FLOW_OK)
|
|
goto preroll_error;
|
|
|
|
size = gst_buffer_get_size (buf);
|
|
|
|
if (len != size) {
|
|
GST_INFO_OBJECT (basesink,
|
|
"got different size than requested from sink pad: %u"
|
|
" != %" G_GSIZE_FORMAT, len, size);
|
|
len = MIN (size, len);
|
|
}
|
|
|
|
basesink->segment.position += len;
|
|
|
|
gst_buffer_extract (buf, 0, data, len);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
|
|
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
|
|
|
|
return;
|
|
|
|
error:
|
|
{
|
|
GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
|
|
gst_flow_get_name (ret), ret);
|
|
gst_audio_ring_buffer_pause (rbuf);
|
|
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
|
|
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_audio_base_sink_drain (sink);
|
|
gst_audio_ring_buffer_pause (rbuf);
|
|
gst_element_post_message (GST_ELEMENT_CAST (sink),
|
|
gst_message_new_eos (GST_OBJECT_CAST (sink)));
|
|
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
|
|
return;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we are flushing");
|
|
gst_audio_ring_buffer_pause (rbuf);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
|
|
return;
|
|
}
|
|
preroll_error:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
|
|
gst_audio_ring_buffer_pause (rbuf);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active)
|
|
{
|
|
gboolean ret;
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink);
|
|
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesink, "activating pull");
|
|
|
|
gst_audio_ring_buffer_set_callback (sink->ringbuffer,
|
|
gst_audio_base_sink_callback, sink);
|
|
|
|
ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "deactivating pull");
|
|
gst_audio_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
|
|
ret = gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_audio_base_sink_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:{
|
|
GstAudioRingBuffer *rb;
|
|
|
|
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
|
|
rb = gst_audio_base_sink_create_ringbuffer (sink);
|
|
if (rb == NULL)
|
|
goto create_failed;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->ringbuffer = rb;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
if (!gst_audio_ring_buffer_open_device (sink->ringbuffer)) {
|
|
GST_OBJECT_LOCK (sink);
|
|
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
|
|
sink->ringbuffer = NULL;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
goto open_failed;
|
|
}
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_audio_base_sink_reset_sync (sink);
|
|
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
|
|
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
|
|
|
|
/* Only post clock-provide messages if this is the clock that
|
|
* we've created. If the subclass has overridden it the subclass
|
|
* should post this messages whenever necessary */
|
|
if (gst_audio_base_sink_is_self_provided_clock (sink))
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
|
|
sink->provided_clock, TRUE));
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
{
|
|
gboolean eos;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
|
|
sink->priv->sync_latency = TRUE;
|
|
eos = GST_BASE_SINK (sink)->eos;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
|
|
if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_PAD_MODE_PULL ||
|
|
g_atomic_int_get (&sink->eos_rendering) || eos) {
|
|
/* we always start the ringbuffer in pull mode immediately */
|
|
/* sync rendering on eos needs running clock,
|
|
* and others need running clock when finished rendering eos */
|
|
gst_audio_ring_buffer_start (sink->ringbuffer);
|
|
}
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* ringbuffer cannot start anymore */
|
|
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
|
|
gst_audio_ring_buffer_pause (sink->ringbuffer);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->sync_latency = FALSE;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* Only post clock-lost messages if this is the clock that
|
|
* we've created. If the subclass has overridden it the subclass
|
|
* should post this messages whenever necessary */
|
|
if (gst_audio_base_sink_is_self_provided_clock (sink))
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
|
|
sink->provided_clock));
|
|
|
|
/* make sure we unblock before calling the parent state change
|
|
* so it can grab the STREAM_LOCK */
|
|
gst_audio_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_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
|
|
gst_audio_ring_buffer_release (sink->ringbuffer);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
/* we release again here because the acquire 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_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
|
|
gst_audio_ring_buffer_release (sink->ringbuffer);
|
|
gst_audio_ring_buffer_close_device (sink->ringbuffer);
|
|
GST_OBJECT_LOCK (sink);
|
|
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
|
|
sink->ringbuffer = NULL;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
create_failed:
|
|
{
|
|
/* subclass must post a meaningful error message */
|
|
GST_DEBUG_OBJECT (sink, "create failed");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
open_failed:
|
|
{
|
|
/* subclass must post a meaningful error message */
|
|
GST_DEBUG_OBJECT (sink, "open failed");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|