/* 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; }; /* BaseAudioSink signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; /* we tollerate half a second diff before we start resyncing. This * should be enough to compensate for various rounding errors in the timestamp * and sample offset position. * This is an emergency resync fallback since buffers marked as DISCONT will * always lock to the correct timestamp immediatly and buffers not marked as * DISCONT are contiguous by definition. */ #define DIFF_TOLERANCE 2 /* 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 enum { PROP_0, PROP_BUFFER_TIME, PROP_LATENCY_TIME, PROP_PROVIDE_CLOCK, PROP_SLAVE_METHOD }; #define GST_TYPE_SLAVE_METHOD (slave_method_get_type ()) static GType slave_method_get_type (void) { static GType slave_method_type = 0; static const GEnumValue slave_method[] = { {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "Resampling slaving", "resample"}, {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "Skew slaving", "skew"}, {0, NULL, NULL}, }; if (!slave_method_type) { slave_method_type = g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method); } return slave_method_type; } #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element"); GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_base_audio_sink, GstBaseSink, 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); static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink * basesink); 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 guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */ static void gst_base_audio_sink_base_init (gpointer g_class) { } 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_DEBUG_FUNCPTR (gst_base_audio_sink_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property); gobject_class->dispose = GST_DEBUG_FUNCPTR (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_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_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_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_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, G_PARAM_READWRITE)); 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); gstbasesink_class->async_play = GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play); gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull); } static void gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink, GstBaseAudioSinkClass * g_class) { 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->provided_clock = gst_audio_clock_new ("GstAudioSinkClock", (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink); GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE; /* FIXME: fix state changes so that both READY_TO_PAUSED and PAUSED_TO_PLAYING return SUCCESS */ GST_BASE_SINK (baseaudiosink)->can_activate_pull = TRUE; } static void gst_base_audio_sink_dispose (GObject * object) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (object); if (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 (GstElement * element, GstQuery * query) { gboolean res = FALSE; GstBaseAudioSink *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"); if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) { GST_DEBUG_OBJECT (basesink, "we are not yet negotiated, can't report latency yet"); res = FALSE; goto done; } /* 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; spec = &basesink->ringbuffer->spec; basesink->priv->us_latency = min_l; min_latency = gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND, spec->rate * spec->bytes_per_sample); /* 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 = 0; max_latency = -1; } gst_query_set_latency (query, live, min_latency, max_latency); } 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, us_latency; 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); /* latency before starting the clock */ us_latency = sink->priv->us_latency; result += us_latency; GST_DEBUG_OBJECT (sink, "processed samples: raw %llu, delay %u, real %llu, time %" GST_TIME_FORMAT ", upstream latency %" GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result), GST_TIME_ARGS (us_latency)); 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_OBJECT_LOCK (sink); sink->provide_clock = g_value_get_boolean (value); GST_OBJECT_UNLOCK (sink); break; case PROP_SLAVE_METHOD: sink->priv->slave_method = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_base_audio_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstBaseAudioSink *sink; sink = GST_BASE_AUDIO_SINK (object); switch (prop_id) { case PROP_BUFFER_TIME: g_value_set_int64 (value, sink->buffer_time); break; case PROP_LATENCY_TIME: g_value_set_int64 (value, sink->latency_time); break; case PROP_PROVIDE_CLOCK: GST_OBJECT_LOCK (sink); g_value_set_boolean (value, sink->provide_clock); GST_OBJECT_UNLOCK (sink); break; case PROP_SLAVE_METHOD: g_value_set_enum (value, sink->priv->slave_method); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); GstRingBufferSpec *spec; if (!sink->ringbuffer) return FALSE; spec = &sink->ringbuffer->spec; GST_DEBUG_OBJECT (sink, "release old ringbuffer"); /* release old ringbuffer */ gst_ring_buffer_release (sink->ringbuffer); GST_DEBUG_OBJECT (sink, "parse caps"); spec->buffer_time = sink->buffer_time; spec->latency_time = sink->latency_time; /* parse new caps */ if (!gst_ring_buffer_parse_caps (spec, caps)) goto parse_error; gst_ring_buffer_debug_spec_buff (spec); GST_DEBUG_OBJECT (sink, "acquire new ringbuffer"); if (!gst_ring_buffer_acquire (sink->ringbuffer, spec)) goto acquire_error; /* calculate actual latency and buffer times. * FIXME: In 0.11, store the latency_time internally in ns */ spec->latency_time = gst_util_uint64_scale (spec->segsize, (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); spec->buffer_time = spec->segtotal * spec->latency_time; gst_ring_buffer_debug_spec_buff (spec); return TRUE; /* ERRORS */ parse_error: { GST_DEBUG_OBJECT (sink, "could not parse caps"); GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("cannot parse audio format.")); return FALSE; } acquire_error: { GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer"); return FALSE; } } static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps) { GstStructure *s; gint width, depth; s = gst_caps_get_structure (caps, 0); /* fields for all formats */ gst_structure_fixate_field_nearest_int (s, "rate", 44100); gst_structure_fixate_field_nearest_int (s, "channels", 2); gst_structure_fixate_field_nearest_int (s, "width", 16); /* fields for int */ if (gst_structure_has_field (s, "depth")) { gst_structure_get_int (s, "width", &width); /* round width to nearest multiple of 8 for the depth */ depth = GST_ROUND_UP_8 (width); gst_structure_fixate_field_nearest_int (s, "depth", depth); } if (gst_structure_has_field (s, "signed")) gst_structure_fixate_field_boolean (s, "signed", TRUE); if (gst_structure_has_field (s, "endianness")) gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER); } static void gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* our clock sync is a bit too much for the base class to handle so * we implement it ourselves. */ *start = GST_CLOCK_TIME_NONE; *end = GST_CLOCK_TIME_NONE; } /* FIXME, this waits for the drain to happen but it cannot be * canceled. */ static gboolean gst_base_audio_sink_drain (GstBaseAudioSink * sink) { if (!sink->ringbuffer) return TRUE; if (!sink->ringbuffer->spec.rate) return TRUE; /* need to start playback before we can drain, but only when * we have successfully negotiated a format and thus aqcuired the * ringbuffer. */ if (gst_ring_buffer_is_acquired (sink->ringbuffer)) gst_ring_buffer_start (sink->ringbuffer); if (sink->next_sample != -1) { GstClockTime time; GstClock *clock; time = gst_util_uint64_scale_int (sink->next_sample, GST_SECOND, sink->ringbuffer->spec.rate); GST_OBJECT_LOCK (sink); if ((clock = GST_ELEMENT_CLOCK (sink)) != NULL) { GstClockID id = gst_clock_new_single_shot_id (clock, time); GST_OBJECT_UNLOCK (sink); GST_DEBUG_OBJECT (sink, "waiting for last sample to play"); gst_clock_id_wait (id, NULL); gst_clock_id_unref (id); sink->next_sample = -1; } else { GST_OBJECT_UNLOCK (sink); } } return TRUE; } static gboolean gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: if (sink->ringbuffer) gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE); break; case GST_EVENT_FLUSH_STOP: /* always resync on sample after a flush */ sink->priv->avg_skew = -1; sink->next_sample = -1; if (sink->ringbuffer) gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); break; case GST_EVENT_EOS: /* now wait till we played everything */ gst_base_audio_sink_drain (sink); break; case GST_EVENT_NEWSEGMENT: { gdouble rate; /* we only need the rate */ gst_event_parse_new_segment_full (event, NULL, &rate, NULL, NULL, NULL, NULL, NULL); GST_DEBUG_OBJECT (sink, "new segment rate of %f", rate); break; } default: break; } return TRUE; } static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer) { GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); if (!gst_ring_buffer_is_acquired (sink->ringbuffer)) goto wrong_state; /* we don't really do anything when prerolling. We could make a * property to play this buffer to have some sort of scrubbing * support. */ return GST_FLOW_OK; wrong_state: { GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state"); GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated.")); return GST_FLOW_NOT_NEGOTIATED; } } static guint64 gst_base_audio_sink_get_offset (GstBaseAudioSink * sink) { guint64 sample; gint writeseg, segdone, sps; gint diff; /* assume we can append to the previous sample */ sample = sink->next_sample; /* no previous sample, try to insert at position 0 */ if (sample == -1) sample = 0; sps = sink->ringbuffer->samples_per_seg; /* figure out the segment and the offset inside the segment where * the sample should be written. */ writeseg = sample / sps; /* get the currently processed segment */ segdone = g_atomic_int_get (&sink->ringbuffer->segdone) - sink->ringbuffer->segbase; /* see how far away it is from the write segment */ diff = writeseg - segdone; if (diff < 0) { /* sample would be dropped, position to next playable position */ sample = (segdone + 1) * sps; } return sample; } static GstClockTime clock_convert_external (GstClockTime external, GstClockTime cinternal, GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom, GstClockTime us_latency) { /* adjust for rate and speed */ if (external >= cexternal) { external = gst_util_uint64_scale (external - cexternal, crate_denom, crate_num); external += cinternal; } else { external = gst_util_uint64_scale (cexternal - external, crate_denom, crate_num); if (cinternal > external) external = cinternal - external; else external = 0; } /* adjust for offset when slaving started */ if (external > us_latency) external -= us_latency; else external = 0; return external; } /* algorithm to calculate sample positions that will result in resampling to * match the clock rate of the master */ static void gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { GstClockTime cinternal, cexternal; GstClockTime crate_num, crate_denom; /* get calibration parameters to compensate for speed and offset differences * when we are slaved */ gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal, &crate_num, &crate_denom); GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f", GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num, crate_denom, gst_guint64_to_gdouble (crate_num) / gst_guint64_to_gdouble (crate_denom)); if (crate_num == 0) crate_denom = crate_num = 1; /* bring external time to internal time */ render_start = clock_convert_external (render_start, cinternal, cexternal, crate_num, crate_denom, sink->priv->us_latency); render_stop = clock_convert_external (render_stop, cinternal, cexternal, crate_num, crate_denom, sink->priv->us_latency); GST_DEBUG_OBJECT (sink, "after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); *srender_start = render_start; *srender_stop = render_stop; } /* algorithm to calculate sample positions that will result in changing the * playout pointer to match the clock rate of the master */ static void gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { GstClockTime cinternal, cexternal, crate_num, crate_denom; GstClockTime etime, itime; GstClockTimeDiff skew, segtime; /* get calibration parameters to compensate for offsets */ gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal, &crate_num, &crate_denom); /* sample clocks and figure out clock skew */ etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink)); itime = gst_clock_get_internal_time (sink->provided_clock); etime -= cexternal; itime -= cinternal; skew = GST_CLOCK_DIFF (etime, itime); if (sink->priv->avg_skew == -1) { /* first observation */ sink->priv->avg_skew = skew; } else { /* next observations use a moving average */ sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32; } GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT, GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew); /* the max drift we allow is the length of a segment */ segtime = sink->ringbuffer->spec.latency_time * 1000; /* adjust playout pointer based on skew */ if (sink->priv->avg_skew > segtime) { /* 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, segtime); cinternal += segtime; sink->priv->avg_skew -= segtime; sink->next_sample = -1; gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal, crate_num, crate_denom); } else if (sink->priv->avg_skew < -segtime) { /* 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, -segtime); cexternal += segtime; sink->priv->avg_skew += segtime; sink->next_sample = -1; gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal, crate_num, crate_denom); } /* convert, ignoring speed */ render_start = clock_convert_external (render_start, cinternal, cexternal, crate_num, crate_denom, sink->priv->us_latency); render_stop = clock_convert_external (render_stop, cinternal, cexternal, crate_num, crate_denom, sink->priv->us_latency); *srender_start = render_start; *srender_stop = render_stop; } /* 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; default: g_warning ("unknown slaving method %d", sink->priv->slave_method); break; } } static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) { guint64 in_offset; GstClockTime time, stop, render_start, render_stop, sample_offset; GstBaseAudioSink *sink; GstRingBuffer *ringbuf; gint64 diff, align, ctime, cstop; guint8 *data; guint size; guint samples, written; gint bps; gint accum; gint out_samples; GstClockTime base_time = -1, latency; GstClock *clock; gboolean sync, slaved, align_next; sink = GST_BASE_AUDIO_SINK (bsink); ringbuf = sink->ringbuffer; /* can't do anything when we don't have the device */ if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf))) goto wrong_state; bps = ringbuf->spec.bytes_per_sample; size = GST_BUFFER_SIZE (buf); if (G_UNLIKELY (size % bps) != 0) goto wrong_size; samples = size / bps; out_samples = samples; in_offset = GST_BUFFER_OFFSET (buf); time = GST_BUFFER_TIMESTAMP (buf); stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, ringbuf->spec.rate); GST_DEBUG_OBJECT (sink, "time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment.start), samples); data = GST_BUFFER_DATA (buf); /* if not valid timestamp or we can't clip or sync, try to play * sample ASAP */ if (!GST_CLOCK_TIME_IS_VALID (time)) { render_start = gst_base_audio_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT, GST_BUFFER_SIZE (buf), render_start); goto no_sync; } /* samples should be rendered based on their timestamp. All samples * arriving before the segment.start or after segment.stop are to be * thrown away. All samples should also be clipped to the segment * boundaries */ /* let's calc stop based on the number of samples in the buffer instead * of trusting the DURATION */ if (!gst_segment_clip (&bsink->segment, GST_FORMAT_TIME, time, stop, &ctime, &cstop)) goto out_of_segment; /* see if some clipping happened */ diff = ctime - time; if (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff); samples -= diff; data += diff * bps; time = ctime; } diff = stop - cstop; if (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff); samples -= diff; stop = cstop; } /* figure out how to sync */ if ((clock = GST_ELEMENT_CLOCK (bsink))) sync = bsink->sync; else sync = FALSE; if (!sync) { /* no sync needed, play sample ASAP */ render_start = gst_base_audio_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start); goto no_sync; } /* bring buffer start and stop times to running time */ render_start = gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time); render_stop = gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop); GST_DEBUG_OBJECT (sink, "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink)); GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT, GST_TIME_ARGS (base_time)); /* add base time to sync against the clock */ render_start += base_time; render_stop += base_time; /* compensate for latency */ latency = gst_base_sink_get_latency (bsink); GST_DEBUG_OBJECT (sink, "compensating for latency %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); /* add latency to get the timestamp to sync against the pipeline clock */ render_start += latency; render_stop += latency; GST_DEBUG_OBJECT (sink, "after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); slaved = clock != sink->provided_clock; if (slaved) { /* handle clock slaving */ gst_base_audio_sink_handle_slaving (sink, render_start, render_stop, &render_start, &render_stop); } /* and bring the time to the rate corrected offset in the buffer */ render_start = gst_util_uint64_scale_int (render_start, ringbuf->spec.rate, GST_SECOND); render_stop = gst_util_uint64_scale_int (render_stop, ringbuf->spec.rate, GST_SECOND); /* always resync after a discont */ if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) { GST_DEBUG_OBJECT (sink, "resync after discont"); goto no_align; } if (G_UNLIKELY (sink->next_sample == -1)) { GST_DEBUG_OBJECT (sink, "no align possible: no previous sample position known"); goto no_align; } /* positive playback rate, first sample is render_start, negative rate, first * sample is render_stop */ if (bsink->segment.rate >= 1.0) sample_offset = render_start; else sample_offset = render_stop; /* now try to align the sample to the previous one */ if (sample_offset >= sink->next_sample) diff = sample_offset - sink->next_sample; else diff = sink->next_sample - sample_offset; /* we tollerate half a second diff before we start resyncing. This * should be enough to compensate for various rounding errors in the timestamp * and sample offset position. We always resync if we got a discont anyway and * non-discont should be aligned by definition. */ if (G_LIKELY (diff < ringbuf->spec.rate / DIFF_TOLERANCE)) { GST_DEBUG_OBJECT (sink, "align with prev sample, %" G_GINT64_FORMAT " < %d", diff, ringbuf->spec.rate / DIFF_TOLERANCE); /* calc align with previous sample */ align = sink->next_sample - sample_offset; } else { /* bring sample diff to seconds for error message */ diff = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate); /* timestamps drifted apart from previous samples too much, we need to * resync. We log this as an element warning. */ GST_ELEMENT_WARNING (sink, CORE, CLOCK, ("Compensating for audio synchronisation problems"), ("Unexpected discontinuity in audio timestamps of more " "than half a second (%" GST_TIME_FORMAT "), resyncing", GST_TIME_ARGS (diff))); align = 0; } /* apply alignment */ render_start += align; /* only align stop if we are not slaved to resample */ if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) { GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved"); goto no_align; } render_stop += align; no_align: /* number of target samples is difference between start and stop */ out_samples = render_stop - render_start; no_sync: /* we render the first or last sample first, depending on the rate */ if (bsink->segment.rate >= 1.0) sample_offset = render_start; else sample_offset = render_stop; GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d", sample_offset, samples, out_samples); /* we need to accumulate over different runs for when we get interrupted */ accum = 0; align_next = TRUE; do { written = gst_ring_buffer_commit_full (ringbuf, &sample_offset, data, samples, out_samples, &accum); GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples); /* if we wrote all, we're done */ if (written == samples) break; /* else something interrupted us and we wait for preroll. */ if (gst_base_sink_wait_preroll (bsink) != GST_FLOW_OK) goto stopping; /* if we got interrupted, we cannot assume that the next sample should * be aligned to this one */ align_next = FALSE; samples -= written; data += written * bps; } while (TRUE); if (align_next) sink->next_sample = sample_offset; else sink->next_sample = -1; GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT, sink->next_sample); if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) { GST_DEBUG_OBJECT (sink, "start playback because we are at the end of segment"); gst_ring_buffer_start (ringbuf); } return GST_FLOW_OK; /* SPECIAL cases */ out_of_segment: { GST_DEBUG_OBJECT (sink, "dropping sample out of segment time %" GST_TIME_FORMAT ", start %" GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (bsink->segment.start)); return GST_FLOW_OK; } /* ERRORS */ wrong_state: { GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated"); GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated.")); return GST_FLOW_NOT_NEGOTIATED; } wrong_size: { GST_DEBUG_OBJECT (sink, "wrong size"); GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE, (NULL), ("sink received buffer of wrong size.")); return GST_FLOW_ERROR; } stopping: { GST_DEBUG_OBJECT (sink, "ringbuffer is stopping"); return GST_FLOW_WRONG_STATE; } } /** * gst_base_audio_sink_create_ringbuffer: * @sink: a #GstBaseAudioSink. * * Create and return the #GstRingBuffer for @sink. This function will call the * ::create_ringbuffer vmethod and will set @sink as the parent of the returned * buffer (see gst_object_set_parent()). * * Returns: The new ringbuffer of @sink. */ GstRingBuffer * gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) { GstBaseAudioSinkClass *bclass; GstRingBuffer *buffer = NULL; bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink); if (bclass->create_ringbuffer) buffer = bclass->create_ringbuffer (sink); if (buffer) gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink)); return buffer; } static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active) { gboolean ret; GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink); if (active) { gst_ring_buffer_set_callback (sink->ringbuffer, gst_base_audio_sink_callback, sink); ret = gst_ring_buffer_start (sink->ringbuffer); } else { gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL); /* stop thread */ ret = gst_ring_buffer_release (sink->ringbuffer); } return ret; } static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len, gpointer user_data) { GstBaseSink *basesink; GstBaseAudioSink *sink; GstBuffer *buf; GstFlowReturn ret; basesink = GST_BASE_SINK (user_data); sink = GST_BASE_AUDIO_SINK (user_data); /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we will copy twice, once into data, once into DMA */ GST_LOG_OBJECT (basesink, "pulling %d bytes offset %" G_GUINT64_FORMAT " to fill audio buffer", len, basesink->offset); ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf); if (ret != GST_FLOW_OK) goto error; if (len != GST_BUFFER_SIZE (buf)) { GST_INFO_OBJECT (basesink, "short read pulling from sink pad: %d<%d", len, GST_BUFFER_SIZE (buf)); len = MIN (GST_BUFFER_SIZE (buf), len); } basesink->offset += len; memcpy (data, GST_BUFFER_DATA (buf), len); return; error: { GST_WARNING_OBJECT (basesink, "Got flow error but can't return it: %d", ret); return; } } /* should be called with the LOCK */ static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink * basesink) { GstClock *clock; GstBaseAudioSink *sink; GstClockTime itime, etime; GstClockTime rate_num, rate_denom; sink = GST_BASE_AUDIO_SINK (basesink); GST_DEBUG_OBJECT (sink, "ringbuffer may start now"); gst_ring_buffer_may_start (sink->ringbuffer, TRUE); clock = GST_ELEMENT_CLOCK (sink); if (clock == NULL) goto done; /* we provided the global clock, don't need to do anything special */ if (clock == sink->provided_clock) goto done; /* if we are slaved to a clock, we need to set the initial * calibration */ /* get external and internal time to set as calibration params */ etime = gst_clock_get_time (clock); itime = gst_clock_get_internal_time (sink->provided_clock); sink->priv->avg_skew = -1; GST_DEBUG_OBJECT (sink, "internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT, GST_TIME_ARGS (itime), GST_TIME_ARGS (etime)); gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num, &rate_denom); gst_clock_set_calibration (sink->provided_clock, itime, etime, rate_num, rate_denom); switch (sink->priv->slave_method) { case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: /* only set as master if we need to resample */ GST_DEBUG_OBJECT (sink, "Setting clock as master"); gst_clock_set_master (sink->provided_clock, clock); break; default: break; } /* start ringbuffer so we can start slaving right away when we need to */ gst_ring_buffer_start (sink->ringbuffer); done: return GST_STATE_CHANGE_SUCCESS; } static GstStateChangeReturn gst_base_audio_sink_do_play (GstBaseAudioSink * sink) { GstStateChangeReturn ret; GST_OBJECT_LOCK (sink); ret = gst_base_audio_sink_async_play (GST_BASE_SINK_CAST (sink)); GST_OBJECT_UNLOCK (sink); return ret; } static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (sink->ringbuffer == NULL) { sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink); } if (!gst_ring_buffer_open_device (sink->ringbuffer)) goto open_failed; break; case GST_STATE_CHANGE_READY_TO_PAUSED: sink->next_sample = -1; gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); gst_ring_buffer_may_start (sink->ringbuffer, FALSE); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: gst_base_audio_sink_do_play (sink); break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* need to take the lock so we don't interfere with an * async play */ GST_OBJECT_LOCK (sink); /* ringbuffer cannot start anymore */ gst_ring_buffer_may_start (sink->ringbuffer, FALSE); gst_ring_buffer_pause (sink->ringbuffer); GST_OBJECT_UNLOCK (sink); break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* make sure we unblock before calling the parent state change * so it can grab the STREAM_LOCK */ gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* stop slaving ourselves to the master, if any */ gst_clock_set_master (sink->provided_clock, NULL); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_ring_buffer_release (sink->ringbuffer); break; case GST_STATE_CHANGE_READY_TO_NULL: gst_ring_buffer_close_device (sink->ringbuffer); break; default: break; } return ret; /* ERRORS */ open_failed: { /* subclass must post a meaningfull error message */ GST_DEBUG_OBJECT (sink, "open failed"); return GST_STATE_CHANGE_FAILURE; } }