/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2005 Wim Taymans * * gstbaseaudiosink.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:gstbaseaudiosink * @short_description: Base class for audio sinks * @see_also: #GstAudioSink, #GstRingBuffer. * * This is the base class for audio sinks. Subclasses need to implement the * ::create_ringbuffer vmethod. This base class will then take care of * writing samples to the ringbuffer, synchronisation, clipping and flushing. * * Last reviewed on 2006-09-27 (0.10.12) */ #include #include "gstbaseaudiosink.h" GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug); #define GST_CAT_DEFAULT gst_base_audio_sink_debug #define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate)) struct _GstBaseAudioSinkPrivate { /* upstream latency */ GstClockTime us_latency; /* the clock slaving algorithm in use */ GstBaseAudioSinkSlaveMethod slave_method; /* running average of clock skew */ GstClockTimeDiff avg_skew; /* the number of samples we aligned last time */ gint64 last_align; gboolean sync_latency; GstClockTime eos_time; /* number of microseconds we alow timestamps or clock slaving to drift * before resyncing */ guint64 drift_tolerance; }; /* BaseAudioSink signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; /* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */ #define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND) #define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND) #define DEFAULT_PROVIDE_CLOCK TRUE #define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SINK_SLAVE_SKEW /* FIXME, enable pull mode when clock slaving and trick modes are figured out */ #define DEFAULT_CAN_ACTIVATE_PULL FALSE /* when timestamps or clock slaving drift for more than 40ms we resync. This is * a reasonable default */ #define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND) enum { PROP_0, PROP_BUFFER_TIME, PROP_LATENCY_TIME, PROP_PROVIDE_CLOCK, PROP_SLAVE_METHOD, PROP_CAN_ACTIVATE_PULL, PROP_DRIFT_TOLERANCE, PROP_LAST }; GType gst_base_audio_sink_slave_method_get_type (void) { static volatile gsize slave_method_type = 0; static const GEnumValue slave_method[] = { {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE", "resample"}, {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "GST_BASE_AUDIO_SINK_SLAVE_SKEW", "skew"}, {GST_BASE_AUDIO_SINK_SLAVE_NONE, "GST_BASE_AUDIO_SINK_SLAVE_NONE", "none"}, {0, NULL, NULL}, }; if (g_once_init_enter (&slave_method_type)) { GType tmp = g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method); g_once_init_leave (&slave_method_type, tmp); } return (GType) slave_method_type; } #define _do_init \ GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element"); #define gst_base_audio_sink_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSink, gst_base_audio_sink, GST_TYPE_BASE_SINK, _do_init); static void gst_base_audio_sink_dispose (GObject * object); static void gst_base_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_base_audio_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); #if 0 static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink * basesink); #endif static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement * element, GstStateChange transition); static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active); static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery * query); static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem); static GstClockTime gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink); static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len, gpointer user_data); static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer); static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer); static gboolean gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event); static void gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps); static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps); static gboolean gst_base_audio_sink_query_pad (GstPad * pad, GstQuery * query); /* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */ static void gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate)); gobject_class->set_property = gst_base_audio_sink_set_property; gobject_class->get_property = gst_base_audio_sink_get_property; gobject_class->dispose = gst_base_audio_sink_dispose; g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_int64 ("buffer-time", "Buffer Time", "Size of audio buffer in microseconds", 1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LATENCY_TIME, g_param_spec_int64 ("latency-time", "Latency Time", "Audio latency in microseconds", 1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK, g_param_spec_boolean ("provide-clock", "Provide Clock", "Provide a clock to be used as the global pipeline clock", DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD, g_param_spec_enum ("slave-method", "Slave Method", "Algorithm to use to match the rate of the masterclock", GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL, g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling", "Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseAudioSink:drift-tolerance * * Controls the amount of time in milliseconds that timestamps or clocks are allowed * to drift before resynchronisation happens. * * Since: 0.10.26 */ g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE, g_param_spec_int64 ("drift-tolerance", "Drift Tolerance", "Tolerance for timestamp and clock drift in microseconds", 1, G_MAXINT64, DEFAULT_DRIFT_TOLERANCE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state); gstelement_class->provide_clock = GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock); gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query); gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event); gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll); gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render); gstbasesink_class->get_times = GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times); gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps); gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate); #if 0 gstbasesink_class->async_play = GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play); #endif gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull); /* ref class from a thread-safe context to work around missing bit of * thread-safety in GObject */ g_type_class_ref (GST_TYPE_AUDIO_CLOCK); g_type_class_ref (GST_TYPE_RING_BUFFER); } static void gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink) { GstBaseSink *basesink; baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink); baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME; baseaudiosink->latency_time = DEFAULT_LATENCY_TIME; baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK; baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD; baseaudiosink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE; baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock", (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink); basesink = GST_BASE_SINK_CAST (baseaudiosink); basesink->can_activate_push = TRUE; basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL; gst_base_sink_set_last_buffer_enabled (basesink, FALSE); /* install some custom pad_query functions */ gst_pad_set_query_function (GST_BASE_SINK_PAD (baseaudiosink), GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad)); } static void gst_base_audio_sink_dispose (GObject * object) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (object); if (sink->provided_clock) { gst_audio_clock_invalidate (sink->provided_clock); gst_object_unref (sink->provided_clock); sink->provided_clock = NULL; } if (sink->ringbuffer) { gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer)); sink->ringbuffer = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static GstClock * gst_base_audio_sink_provide_clock (GstElement * elem) { GstBaseAudioSink *sink; GstClock *clock; sink = GST_BASE_AUDIO_SINK (elem); /* we have no ringbuffer (must be NULL state) */ if (sink->ringbuffer == NULL) goto wrong_state; if (!gst_ring_buffer_is_acquired (sink->ringbuffer)) goto wrong_state; GST_OBJECT_LOCK (sink); if (!sink->provide_clock) goto clock_disabled; clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock)); GST_OBJECT_UNLOCK (sink); return clock; /* ERRORS */ wrong_state: { GST_DEBUG_OBJECT (sink, "ringbuffer not acquired"); return NULL; } clock_disabled: { GST_DEBUG_OBJECT (sink, "clock provide disabled"); GST_OBJECT_UNLOCK (sink); return NULL; } } static gboolean gst_base_audio_sink_query_pad (GstPad * pad, GstQuery * query) { gboolean res = FALSE; GstBaseAudioSink *basesink; basesink = GST_BASE_AUDIO_SINK (gst_pad_get_parent (pad)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CONVERT: { GstFormat src_fmt, dest_fmt; gint64 src_val, dest_val; GST_LOG_OBJECT (pad, "query convert"); if (basesink->ringbuffer) { gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL); res = gst_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: break; } gst_object_unref (basesink); return res; } static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery * query) { gboolean res = FALSE; GstBaseAudioSink *basesink; basesink = GST_BASE_AUDIO_SINK (element); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { gboolean live, us_live; GstClockTime min_l, max_l; GST_DEBUG_OBJECT (basesink, "latency query"); /* ask parent first, it will do an upstream query for us. */ if ((res = gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live, &us_live, &min_l, &max_l))) { GstClockTime min_latency, max_latency; /* we and upstream are both live, adjust the min_latency */ if (live && us_live) { GstRingBufferSpec *spec; GST_OBJECT_LOCK (basesink); if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) { GST_OBJECT_UNLOCK (basesink); GST_DEBUG_OBJECT (basesink, "we are not yet negotiated, can't report latency yet"); res = FALSE; goto done; } spec = &basesink->ringbuffer->spec; basesink->priv->us_latency = min_l; min_latency = gst_util_uint64_scale_int (spec->seglatency * spec->segsize, GST_SECOND, spec->rate * spec->bytes_per_sample); GST_OBJECT_UNLOCK (basesink); /* we cannot go lower than the buffer size and the min peer latency */ min_latency = min_latency + min_l; /* the max latency is the max of the peer, we can delay an infinite * amount of time. */ max_latency = min_latency + (max_l == -1 ? 0 : 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)); } 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_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; } static GstClockTime gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink) { guint64 raw, samples; guint delay; GstClockTime result; if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0) return GST_CLOCK_TIME_NONE; /* our processed samples are always increasing */ raw = samples = gst_ring_buffer_samples_done (sink->ringbuffer); /* the number of samples not yet processed, this is still queued in the * device (not played for playback). */ delay = gst_ring_buffer_delay (sink->ringbuffer); if (G_LIKELY (samples >= delay)) samples -= delay; else samples = 0; result = gst_util_uint64_scale_int (samples, GST_SECOND, sink->ringbuffer->spec.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_base_audio_sink_set_provide_clock: * @sink: a #GstBaseAudioSink * @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. * * Since: 0.10.16 */ void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink, gboolean provide) { g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); GST_OBJECT_LOCK (sink); sink->provide_clock = provide; GST_OBJECT_UNLOCK (sink); } /** * gst_base_audio_sink_get_provide_clock: * @sink: a #GstBaseAudioSink * * Queries whether @sink will provide a clock or not. See also * gst_base_audio_sink_set_provide_clock. * * Returns: %TRUE if @sink will provide a clock. * * Since: 0.10.16 */ gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink) { gboolean result; g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE); GST_OBJECT_LOCK (sink); result = sink->provide_clock; GST_OBJECT_UNLOCK (sink); return result; } /** * gst_base_audio_sink_set_slave_method: * @sink: a #GstBaseAudioSink * @method: the new slave method * * Controls how clock slaving will be performed in @sink. * * Since: 0.10.16 */ void gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink, GstBaseAudioSinkSlaveMethod method) { g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->slave_method = method; GST_OBJECT_UNLOCK (sink); } /** * gst_base_audio_sink_get_slave_method: * @sink: a #GstBaseAudioSink * * Get the current slave method used by @sink. * * Returns: The current slave method used by @sink. * * Since: 0.10.16 */ GstBaseAudioSinkSlaveMethod gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink) { GstBaseAudioSinkSlaveMethod result; g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->slave_method; GST_OBJECT_UNLOCK (sink); return result; } /** * gst_base_audio_sink_set_drift_tolerance: * @sink: a #GstBaseAudioSink * @drift_tolerance: the new drift tolerance in microseconds * * Controls the sink's drift tolerance. * * Since: 0.10.31 */ void gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink, gint64 drift_tolerance) { g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->drift_tolerance = drift_tolerance; GST_OBJECT_UNLOCK (sink); } /** * gst_base_audio_sink_get_drift_tolerance * @sink: a #GstBaseAudioSink * * Get the current drift tolerance, in microseconds, used by @sink. * * Returns: The current drift tolerance used by @sink. * * Since: 0.10.31 */ gint64 gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink) { gint64 result; g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->drift_tolerance; GST_OBJECT_UNLOCK (sink); return result; } static void gst_base_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (object); switch (prop_id) { case PROP_BUFFER_TIME: sink->buffer_time = g_value_get_int64 (value); break; case PROP_LATENCY_TIME: sink->latency_time = g_value_get_int64 (value); break; case PROP_PROVIDE_CLOCK: gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value)); break; case PROP_SLAVE_METHOD: gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value)); break; case PROP_CAN_ACTIVATE_PULL: GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value); break; case PROP_DRIFT_TOLERANCE: gst_base_audio_sink_set_drift_tolerance (sink, g_value_get_int64 (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_base_audio_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (object); switch (prop_id) { case PROP_BUFFER_TIME: g_value_set_int64 (value, sink->buffer_time); break; case PROP_LATENCY_TIME: g_value_set_int64 (value, sink->latency_time); break; case PROP_PROVIDE_CLOCK: g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink)); break; case PROP_SLAVE_METHOD: g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink)); break; 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_base_audio_sink_get_drift_tolerance (sink)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); GstRingBufferSpec *spec; GstClockTime now; GstClockTime crate_num, crate_denom; if (!sink->ringbuffer) return FALSE; spec = &sink->ringbuffer->spec; 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); GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now)); /* release old ringbuffer */ gst_ring_buffer_pause (sink->ringbuffer); gst_ring_buffer_activate (sink->ringbuffer, FALSE); gst_ring_buffer_release (sink->ringbuffer); GST_DEBUG_OBJECT (sink, "parse caps"); spec->buffer_time = sink->buffer_time; spec->latency_time = sink->latency_time; /* parse new caps */ if (!gst_ring_buffer_parse_caps (spec, caps)) goto parse_error; gst_ring_buffer_debug_spec_buff (spec); GST_DEBUG_OBJECT (sink, "acquire ringbuffer"); if (!gst_ring_buffer_acquire (sink->ringbuffer, spec)) goto acquire_error; if (bsink->pad_mode == GST_ACTIVATE_PUSH) { GST_DEBUG_OBJECT (sink, "activate ringbuffer"); gst_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, gst_clock_get_internal_time (sink->provided_clock), now, crate_num, crate_denom); /* calculate actual latency and buffer times. * FIXME: In 0.11, store the latency_time internally in ns */ spec->latency_time = gst_util_uint64_scale (spec->segsize, (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); spec->buffer_time = spec->segtotal * spec->latency_time; gst_ring_buffer_debug_spec_buff (spec); return TRUE; /* ERRORS */ parse_error: { GST_DEBUG_OBJECT (sink, "could not parse caps"); GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("cannot parse audio format.")); return FALSE; } acquire_error: { GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer"); return FALSE; } } static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps) { GstStructure *s; gint width, depth; s = gst_caps_get_structure (caps, 0); /* fields for all formats */ gst_structure_fixate_field_nearest_int (s, "rate", 44100); gst_structure_fixate_field_nearest_int (s, "channels", 2); gst_structure_fixate_field_nearest_int (s, "width", 16); /* fields for int */ if (gst_structure_has_field (s, "depth")) { gst_structure_get_int (s, "width", &width); /* round width to nearest multiple of 8 for the depth */ depth = GST_ROUND_UP_8 (width); gst_structure_fixate_field_nearest_int (s, "depth", depth); } if (gst_structure_has_field (s, "signed")) gst_structure_fixate_field_boolean (s, "signed", TRUE); if (gst_structure_has_field (s, "endianness")) gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER); } static void gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* our clock sync is a bit too much for the base class to handle so * we implement it ourselves. */ *start = GST_CLOCK_TIME_NONE; *end = GST_CLOCK_TIME_NONE; } /* This waits for the drain to happen and can be canceled */ static gboolean gst_base_audio_sink_drain (GstBaseAudioSink * sink) { if (!sink->ringbuffer) return TRUE; if (!sink->ringbuffer->spec.rate) return TRUE; /* 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_ring_buffer_is_acquired (sink->ringbuffer)) gst_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. */ gst_base_sink_wait_eos (GST_BASE_SINK (sink), sink->priv->eos_time, NULL); GST_DEBUG_OBJECT (sink, "drained audio"); } g_atomic_int_set (&sink->eos_rendering, 0); return TRUE; } static gboolean gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: if (sink->ringbuffer) gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE); break; case GST_EVENT_FLUSH_STOP: /* always resync on sample after a flush */ sink->priv->avg_skew = -1; sink->next_sample = -1; sink->priv->eos_time = -1; if (sink->ringbuffer) gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); break; case GST_EVENT_EOS: /* now wait till we played everything */ gst_base_audio_sink_drain (sink); break; default: break; } return TRUE; } static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); if (!gst_ring_buffer_is_acquired (sink->ringbuffer)) goto wrong_state; /* we don't really do anything when prerolling. We could make a * property to play this buffer to have some sort of scrubbing * support. */ return GST_FLOW_OK; wrong_state: { GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state"); GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated.")); return GST_FLOW_NOT_NEGOTIATED; } } static guint64 gst_base_audio_sink_get_offset (GstBaseAudioSink * sink) { guint64 sample; gint writeseg, segdone, sps; gint diff; /* assume we can append to the previous sample */ sample = sink->next_sample; /* no previous sample, try to insert at position 0 */ if (sample == -1) sample = 0; sps = sink->ringbuffer->samples_per_seg; /* figure out the segment and the offset inside the segment where * the sample should be written. */ writeseg = sample / sps; /* get the currently processed segment */ segdone = g_atomic_int_get (&sink->ringbuffer->segdone) - sink->ringbuffer->segbase; /* see how far away it is from the write segment */ diff = writeseg - segdone; if (diff < 0) { /* sample would be dropped, position to next playable position */ sample = (segdone + 1) * sps; } return sample; } static GstClockTime clock_convert_external (GstClockTime external, GstClockTime cinternal, GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom) { /* 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; } /* algorithm to calculate sample positions that will result in resampling to * match the clock rate of the master */ static void gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { GstClockTime cinternal, cexternal; GstClockTime crate_num, crate_denom; /* 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 comming 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_base_audio_sink_skew_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { GstClockTime cinternal, cexternal, crate_num, crate_denom; GstClockTime etime, itime; GstClockTimeDiff skew, mdrift, 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 (sink->provided_clock); itime = gst_audio_clock_adjust (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 %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT, GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew); /* the max drift we allow */ mdrift = sink->priv->drift_tolerance * 1000; mdrift2 = mdrift / 2; /* adjust playout pointer based on skew */ if (sink->priv->avg_skew > mdrift2) { /* master is running slower, move internal time forward */ GST_WARNING_OBJECT (sink, "correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT, sink->priv->avg_skew, mdrift2); cexternal = cexternal > mdrift ? cexternal - mdrift : 0; sink->priv->avg_skew -= mdrift; driftsamples = (sink->ringbuffer->spec.rate * mdrift) / 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 %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT, sink->priv->avg_skew, -mdrift2); cexternal += mdrift; sink->priv->avg_skew += mdrift; driftsamples = (sink->ringbuffer->spec.rate * mdrift) / 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_base_audio_sink_none_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { GstClockTime cinternal, cexternal, crate_num, crate_denom; /* get calibration parameters to compensate for offsets */ gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal, &crate_num, &crate_denom); /* convert, ignoring speed */ render_start = clock_convert_external (render_start, cinternal, cexternal, crate_num, crate_denom); 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_base_audio_sink_handle_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { switch (sink->priv->slave_method) { case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: gst_base_audio_sink_resample_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; case GST_BASE_AUDIO_SINK_SLAVE_SKEW: gst_base_audio_sink_skew_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; case GST_BASE_AUDIO_SINK_SLAVE_NONE: gst_base_audio_sink_none_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; default: g_warning ("unknown slaving method %d", sink->priv->slave_method); break; } } /* must be called with LOCK */ static GstFlowReturn gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj) { GstClock *clock; GstClockReturn status; GstClockTime time, render_delay; GstFlowReturn ret; GstBaseAudioSink *sink; GstClockTime itime, etime; GstClockTime rate_num, rate_denom; GstClockTimeDiff jitter; sink = GST_BASE_AUDIO_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 occures, we continue. */ } while (status == GST_CLOCK_UNSCHEDULED); GST_OBJECT_LOCK (sink); GST_DEBUG_OBJECT (sink, "latency synced"); /* 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; itime = gst_audio_clock_get_time (sink->provided_clock); itime = gst_audio_clock_adjust (sink->provided_clock, itime); 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_ring_buffer_start (sink->ringbuffer); 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_BASE_AUDIO_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_BASE_AUDIO_SINK_SLAVE_SKEW: case GST_BASE_AUDIO_SINK_SLAVE_NONE: default: break; } sink->priv->avg_skew = -1; sink->next_sample = -1; sink->priv->eos_time = -1; 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_WRONG_STATE; } } static gint64 gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink, GstClockTime sample_offset) { GstRingBuffer *ringbuf = sink->ringbuffer; gint64 align; gint64 diff; gint64 maxdrift; gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase; gint64 samples_done = segdone * ringbuf->samples_per_seg; gint64 headroom = sample_offset - samples_done; gboolean allow_align = TRUE; /* now try to align the sample to the previous one, first see how big the * difference is. */ if (sample_offset >= sink->next_sample) diff = sample_offset - sink->next_sample; else diff = sink->next_sample - sample_offset; /* calculate the max allowed drift in units of samples. By default this is * 20ms and should be anough to compensate for timestamp rounding errors. */ maxdrift = (ringbuf->spec.rate * sink->priv->drift_tolerance) / GST_MSECOND; /* calc align with previous sample */ align = sink->next_sample - sample_offset; /* don't align if it means writing behind the read-segment */ if (diff > headroom && align < 0) allow_align = FALSE; if (G_LIKELY (diff < maxdrift && allow_align)) { GST_DEBUG_OBJECT (sink, "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %" G_GINT64_FORMAT, align, maxdrift); } else { /* calculate sample diff in seconds for error message */ gint64 diff_s = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate); /* timestamps drifted apart from previous samples too much, we need to * resync. We log this as an element warning. */ GST_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; } return align; } static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) { guint64 in_offset; GstClockTime time, stop, render_start, render_stop, sample_offset; GstClockTimeDiff sync_offset, ts_offset; GstBaseAudioSinkClass *bclass; GstBaseAudioSink *sink; GstRingBuffer *ringbuf; gint64 diff, align; guint64 ctime, cstop; gsize offset; guint8 *data; gsize size; guint samples, written; gint bps; 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_BASE_AUDIO_SINK (bsink); bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink); ringbuf = sink->ringbuffer; /* can't do anything when we don't have the device */ if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf))) goto wrong_state; /* 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_base_audio_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; } bps = ringbuf->spec.bytes_per_sample; size = gst_buffer_get_size (buf); if (G_UNLIKELY (size % bps) != 0) goto wrong_size; samples = size / bps; out_samples = samples; in_offset = GST_BUFFER_OFFSET (buf); time = GST_BUFFER_TIMESTAMP (buf); GST_DEBUG_OBJECT (sink, "time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %" GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset, 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_base_audio_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT, size, render_start); /* we don't have a start so we don't know stop either */ stop = -1; goto no_sync; } /* 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, ringbuf->spec.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 %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT ", ts-offset %" G_GINT64_FORMAT, sync_offset, GST_TIME_ARGS (render_delay), ts_offset); /* compensate for ts-offset and device-delay when negative we need to * clip. */ if (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 (!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 (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff); samples -= diff; offset += diff * bps; time = ctime; } diff = stop - cstop; if (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff); samples -= diff; stop = cstop; } /* figure out how to sync */ if ((clock = GST_ELEMENT_CLOCK (bsink))) sync = bsink->sync; else sync = FALSE; if (!sync) { /* no sync needed, play sample ASAP */ render_start = gst_base_audio_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start); goto no_sync; } /* bring buffer start and stop times to running time */ render_start = gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time); render_stop = gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop); GST_DEBUG_OBJECT (sink, "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); /* store the time of the last sample, we'll use this to perform sync on the * last sample when draining the buffer */ if (bsink->segment.rate >= 0.0) { sink->priv->eos_time = render_stop; } else { sink->priv->eos_time = render_start; } /* 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; 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; GST_DEBUG_OBJECT (sink, "after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); if ((slaved = clock != sink->provided_clock)) { /* handle clock slaving */ gst_base_audio_sink_handle_slaving (sink, render_start, render_stop, &render_start, &render_stop); } else { /* no slaving needed but we need to adapt to the clock calibration * parameters */ gst_base_audio_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; GST_DEBUG_OBJECT (sink, "time offset %" GST_TIME_FORMAT, GST_TIME_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; /* and bring the time to the rate corrected offset in the buffer */ render_start = gst_util_uint64_scale_int (render_start, ringbuf->spec.rate, GST_SECOND); render_stop = gst_util_uint64_scale_int (render_stop, ringbuf->spec.rate, GST_SECOND); /* 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 (bsink->segment.rate >= 0.0) { sample_offset = render_start; if (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_DEBUG_OBJECT (sink, "resync after discont"); 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_base_audio_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 (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) { GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved"); goto no_align; } render_stop += align; no_align: /* number of target samples is difference between start and stop */ out_samples = render_stop - render_start; no_sync: /* we render the first or last sample first, depending on the rate */ if (bsink->segment.rate >= 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; data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ); do { written = gst_ring_buffer_commit_full (ringbuf, &sample_offset, data + offset, samples, out_samples, &accum); GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples); /* if we wrote all, we're done */ if (written == samples) break; /* else something interrupted us and we wait for preroll. */ if ((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 * bps; } while (TRUE); gst_buffer_unmap (buf, data, size); if (align_next) sink->next_sample = sample_offset; else sink->next_sample = -1; GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT, sink->next_sample); if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) { GST_DEBUG_OBJECT (sink, "start playback because we are at the end of segment"); gst_ring_buffer_start (ringbuf); } 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; } /* 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, data, size); goto done; } sync_latency_failed: { GST_DEBUG_OBJECT (sink, "failed waiting for latency"); goto done; } } /** * gst_base_audio_sink_create_ringbuffer: * @sink: a #GstBaseAudioSink. * * Create and return the #GstRingBuffer for @sink. This function will call the * ::create_ringbuffer vmethod and will set @sink as the parent of the returned * buffer (see gst_object_set_parent()). * * Returns: The new ringbuffer of @sink. */ GstRingBuffer * gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) { GstBaseAudioSinkClass *bclass; GstRingBuffer *buffer = NULL; bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink); if (bclass->create_ringbuffer) buffer = bclass->create_ringbuffer (sink); if (buffer) gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink)); return buffer; } static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len, gpointer user_data) { GstBaseSink *basesink; GstBaseAudioSink *sink; GstBuffer *buf; GstFlowReturn ret; gsize size; basesink = GST_BASE_SINK (user_data); sink = GST_BASE_AUDIO_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 %d 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_UNEXPECTED) 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 != %u", 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_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_base_audio_sink_drain (sink); gst_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); } flushing: { GST_DEBUG_OBJECT (sink, "we are flushing"); gst_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_ring_buffer_pause (rbuf); GST_BASE_SINK_PREROLL_UNLOCK (basesink); GST_PAD_STREAM_UNLOCK (basesink->sinkpad); return; } } static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active) { gboolean ret; GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink); if (active) { GST_DEBUG_OBJECT (basesink, "activating pull"); gst_ring_buffer_set_callback (sink->ringbuffer, gst_base_audio_sink_callback, sink); ret = gst_ring_buffer_activate (sink->ringbuffer, TRUE); } else { GST_DEBUG_OBJECT (basesink, "deactivating pull"); gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL); ret = gst_ring_buffer_activate (sink->ringbuffer, FALSE); } return ret; } #if 0 /* should be called with the LOCK */ static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink * basesink) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (basesink); GST_DEBUG_OBJECT (sink, "ringbuffer may start now"); sink->priv->sync_latency = TRUE; gst_ring_buffer_may_start (sink->ringbuffer, TRUE); if (basesink->pad_mode == GST_ACTIVATE_PULL) { /* we always start the ringbuffer in pull mode immediatly */ gst_ring_buffer_start (sink->ringbuffer); } return GST_STATE_CHANGE_SUCCESS; } #endif static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (sink->ringbuffer == NULL) { gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0); sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink); } if (!gst_ring_buffer_open_device (sink->ringbuffer)) goto open_failed; break; case GST_STATE_CHANGE_READY_TO_PAUSED: sink->next_sample = -1; sink->priv->last_align = -1; sink->priv->eos_time = -1; gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); gst_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 overriden it the subclass * should post this messages whenever necessary */ if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) && GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func == (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time) 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_ring_buffer_may_start (sink->ringbuffer, TRUE); if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL || g_atomic_int_get (&sink->eos_rendering) || eos) { /* we always start the ringbuffer in pull mode immediatly */ /* sync rendering on eos needs running clock, * and others need running clock when finished rendering eos */ gst_ring_buffer_start (sink->ringbuffer); } break; } case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* ringbuffer cannot start anymore */ gst_ring_buffer_may_start (sink->ringbuffer, FALSE); gst_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 overriden it the subclass * should post this messages whenever necessary */ if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) && GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func == (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time) 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_ring_buffer_set_flushing (sink->ringbuffer, TRUE); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* stop slaving ourselves to the master, if any */ gst_clock_set_master (sink->provided_clock, NULL); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_ring_buffer_activate (sink->ringbuffer, FALSE); gst_ring_buffer_release (sink->ringbuffer); break; case GST_STATE_CHANGE_READY_TO_NULL: /* we release again here because the aqcuire happens when setting the * caps, which happens before we commit the state to PAUSED and thus the * PAUSED->READY state change (see above, where we release the ringbuffer) * might not be called when we get here. */ gst_ring_buffer_activate (sink->ringbuffer, FALSE); gst_ring_buffer_release (sink->ringbuffer); gst_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 */ open_failed: { /* subclass must post a meaningfull error message */ GST_DEBUG_OBJECT (sink, "open failed"); return GST_STATE_CHANGE_FAILURE; } }