/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2001 Thomas * 2005,2006 Wim Taymans * 2013 Sebastian Dröge * * audiomixer.c: AudioMixer element, N in, one out, samples are added * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-audiomixer * * The audiomixer allows to mix several streams into one by adding the data. * Mixed data is clamped to the min/max values of the data format. * * The audiomixer currently mixes all data received on the sinkpads as soon as * possible without trying to synchronize the streams. * * * Example launch line * |[ * gst-launch audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix. * ]| This pipeline produces two sine waves mixed together. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstaudiomixer.h" #include #include /* strcmp */ #include "gstaudiomixerorc.h" #define GST_CAT_DEFAULT gst_audiomixer_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); typedef struct _GstAudioMixerCollect GstAudioMixerCollect; struct _GstAudioMixerCollect { GstCollectData collect; /* we extend the CollectData */ GstBuffer *buffer; /* current buffer we're mixing, for comparison with collect.buffer to see if we need to update our cached values. */ guint position, size; guint64 output_offset; /* Offset in output segment that collect.pos refers to in the current buffer. */ guint64 next_offset; /* Next expected offset in the input segment */ }; #define DEFAULT_PAD_VOLUME (1.0) #define DEFAULT_PAD_MUTE (FALSE) /* some defines for audio processing */ /* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0 * we map 1.0 to VOLUME_UNITY_INT* */ #define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */ #define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */ #define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */ #define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */ #define VOLUME_UNITY_INT32_BIT_SHIFT 27 enum { PROP_PAD_0, PROP_PAD_VOLUME, PROP_PAD_MUTE }; G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad, GST_TYPE_PAD); static void gst_audiomixer_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: g_value_set_double (value, pad->volume); break; case PROP_PAD_MUTE: g_value_set_boolean (value, pad->mute); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: GST_OBJECT_LOCK (pad); pad->volume = g_value_get_double (value); pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8; pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16; pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32; GST_OBJECT_UNLOCK (pad); break; case PROP_PAD_MUTE: GST_OBJECT_LOCK (pad); pad->mute = g_value_get_boolean (value); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->set_property = gst_audiomixer_pad_set_property; gobject_class->get_property = gst_audiomixer_pad_get_property; g_object_class_install_property (gobject_class, PROP_PAD_VOLUME, g_param_spec_double ("volume", "Volume", "Volume of this pad", 0.0, 10.0, DEFAULT_PAD_VOLUME, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PAD_MUTE, g_param_spec_boolean ("mute", "Mute", "Mute this pad", DEFAULT_PAD_MUTE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); } static void gst_audiomixer_pad_init (GstAudioMixerPad * pad) { pad->volume = DEFAULT_PAD_VOLUME; pad->mute = DEFAULT_PAD_MUTE; } #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND) #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) #define DEFAULT_BLOCKSIZE (1024) enum { PROP_0, PROP_FILTER_CAPS, PROP_ALIGNMENT_THRESHOLD, PROP_DISCONT_WAIT, PROP_BLOCKSIZE }; /* elementfactory information */ #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \ ", layout = (string) { interleaved, non-interleaved }" #else #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \ ", layout = (string) { interleaved, non-interleaved }" #endif static GstStaticPadTemplate gst_audiomixer_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (CAPS) ); static GstStaticPadTemplate gst_audiomixer_sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS (CAPS) ); static void gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data); #define gst_audiomixer_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer, GST_TYPE_ELEMENT, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_audiomixer_child_proxy_init)); static void gst_audiomixer_dispose (GObject * object); static void gst_audiomixer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad, GstCaps * caps); static gboolean gst_audiomixer_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_audiomixer_sink_query (GstCollectPads * pads, GstCollectData * pad, GstQuery * query, gpointer user_data); static gboolean gst_audiomixer_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_audiomixer_sink_event (GstCollectPads * pads, GstCollectData * pad, GstEvent * event, gpointer user_data); static GstPad *gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * temp, const gchar * unused, const GstCaps * caps); static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad); static GstStateChangeReturn gst_audiomixer_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_audiomixer_do_clip (GstCollectPads * pads, GstCollectData * data, GstBuffer * buffer, GstBuffer ** out, gpointer user_data); static GstFlowReturn gst_audiomixer_collected (GstCollectPads * pads, gpointer user_data); /* we can only accept caps that we and downstream can handle. * if we have filtercaps set, use those to constrain the target caps. */ static GstCaps * gst_audiomixer_sink_getcaps (GstPad * pad, GstCaps * filter) { GstAudioMixer *audiomixer; GstCaps *result, *peercaps, *current_caps, *filter_caps; GstStructure *s; gint i, n; audiomixer = GST_AUDIO_MIXER (GST_PAD_PARENT (pad)); GST_OBJECT_LOCK (audiomixer); /* take filter */ if ((filter_caps = audiomixer->filter_caps)) { if (filter) filter_caps = gst_caps_intersect_full (filter, filter_caps, GST_CAPS_INTERSECT_FIRST); else gst_caps_ref (filter_caps); } else { filter_caps = filter ? gst_caps_ref (filter) : NULL; } GST_OBJECT_UNLOCK (audiomixer); if (filter_caps && gst_caps_is_empty (filter_caps)) { GST_WARNING_OBJECT (pad, "Empty filter caps"); return filter_caps; } /* get the downstream possible caps */ peercaps = gst_pad_peer_query_caps (audiomixer->srcpad, filter_caps); /* get the allowed caps on this sinkpad */ GST_OBJECT_LOCK (audiomixer); current_caps = audiomixer->current_caps ? gst_caps_ref (audiomixer->current_caps) : NULL; if (current_caps == NULL) { current_caps = gst_pad_get_pad_template_caps (pad); if (!current_caps) current_caps = gst_caps_new_any (); } GST_OBJECT_UNLOCK (audiomixer); if (peercaps) { /* if the peer has caps, intersect */ GST_DEBUG_OBJECT (audiomixer, "intersecting peer and our caps"); result = gst_caps_intersect_full (peercaps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (peercaps); gst_caps_unref (current_caps); } else { /* the peer has no caps (or there is no peer), just use the allowed caps * of this sinkpad. */ /* restrict with filter-caps if any */ if (filter_caps) { GST_DEBUG_OBJECT (audiomixer, "no peer caps, using filtered caps"); result = gst_caps_intersect_full (filter_caps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (current_caps); } else { GST_DEBUG_OBJECT (audiomixer, "no peer caps, using our caps"); result = current_caps; } } result = gst_caps_make_writable (result); n = gst_caps_get_size (result); for (i = 0; i < n; i++) { GstStructure *sref; s = gst_caps_get_structure (result, i); sref = gst_structure_copy (s); gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL); if (gst_structure_is_subset (s, sref)) { /* This field is irrelevant when in mono or stereo */ gst_structure_remove_field (s, "channel-mask"); } gst_structure_free (sref); } if (filter_caps) gst_caps_unref (filter_caps); GST_LOG_OBJECT (audiomixer, "getting caps on pad %p,%s to %" GST_PTR_FORMAT, pad, GST_PAD_NAME (pad), result); return result; } static gboolean gst_audiomixer_sink_query (GstCollectPads * pads, GstCollectData * pad, GstQuery * query, gpointer user_data) { gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_audiomixer_sink_getcaps (pad->pad, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = gst_collect_pads_query_default (pads, pad, query, FALSE); break; } return res; } /* the first caps we receive on any of the sinkpads will define the caps for all * the other sinkpads because we can only mix streams with the same caps. */ static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad, GstCaps * orig_caps) { GstCaps *caps; GstAudioInfo info; GstStructure *s; gint channels; caps = gst_caps_copy (orig_caps); s = gst_caps_get_structure (caps, 0); if (gst_structure_get_int (s, "channels", &channels)) if (channels <= 2) gst_structure_remove_field (s, "channel-mask"); if (!gst_audio_info_from_caps (&info, caps)) goto invalid_format; GST_OBJECT_LOCK (audiomixer); /* don't allow reconfiguration for now; there's still a race between the * different upstream threads doing query_caps + accept_caps + sending * (possibly different) CAPS events, but there's not much we can do about * that, upstream needs to deal with it. */ if (audiomixer->current_caps != NULL) { if (gst_audio_info_is_equal (&info, &audiomixer->info)) { GST_OBJECT_UNLOCK (audiomixer); gst_caps_unref (caps); return TRUE; } else { GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but " "current caps are %" GST_PTR_FORMAT, caps, audiomixer->current_caps); GST_OBJECT_UNLOCK (audiomixer); gst_pad_push_event (pad, gst_event_new_reconfigure ()); gst_caps_unref (caps); return FALSE; } } GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps); gst_caps_replace (&audiomixer->current_caps, caps); memcpy (&audiomixer->info, &info, sizeof (info)); audiomixer->send_caps = TRUE; GST_OBJECT_UNLOCK (audiomixer); /* send caps event later, after stream-start event */ GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps); gst_caps_unref (caps); return TRUE; /* ERRORS */ invalid_format: { gst_caps_unref (caps); GST_WARNING_OBJECT (audiomixer, "invalid format set as caps"); return FALSE; } } /* FIXME, the duration query should reflect how long you will produce * data, that is the amount of stream time until you will emit EOS. * * For synchronized mixing this is always the max of all the durations * of upstream since we emit EOS when all of them finished. * * We don't do synchronized mixing so this really depends on where the * streams where punched in and what their relative offsets are against * eachother which we can get from the first timestamps we see. * * When we add a new stream (or remove a stream) the duration might * also become invalid again and we need to post a new DURATION * message to notify this fact to the parent. * For now we take the max of all the upstream elements so the simple * cases work at least somewhat. */ static gboolean gst_audiomixer_query_duration (GstAudioMixer * audiomixer, GstQuery * query) { gint64 max; gboolean res; GstFormat format; GstIterator *it; gboolean done; GValue item = { 0, }; /* parse format */ gst_query_parse_duration (query, &format, NULL); max = -1; res = TRUE; done = FALSE; it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer)); while (!done) { GstIteratorResult ires; ires = gst_iterator_next (it, &item); switch (ires) { case GST_ITERATOR_DONE: done = TRUE; break; case GST_ITERATOR_OK: { GstPad *pad = g_value_get_object (&item); gint64 duration; /* ask sink peer for duration */ res &= gst_pad_peer_query_duration (pad, format, &duration); /* take max from all valid return values */ if (res) { /* valid unknown length, stop searching */ if (duration == -1) { max = duration; done = TRUE; } /* else see if bigger than current max */ else if (duration > max) max = duration; } g_value_reset (&item); break; } case GST_ITERATOR_RESYNC: max = -1; res = TRUE; gst_iterator_resync (it); break; default: res = FALSE; done = TRUE; break; } } g_value_unset (&item); gst_iterator_free (it); if (res) { /* and store the max */ GST_DEBUG_OBJECT (audiomixer, "Total duration in format %s: %" GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max)); gst_query_set_duration (query, format, max); } return res; } static gboolean gst_audiomixer_query_latency (GstAudioMixer * audiomixer, GstQuery * query) { GstClockTime min, max; gboolean live; gboolean res; GstIterator *it; gboolean done; GValue item = { 0, }; res = TRUE; done = FALSE; live = FALSE; min = 0; max = GST_CLOCK_TIME_NONE; /* Take maximum of all latency values */ it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer)); while (!done) { GstIteratorResult ires; ires = gst_iterator_next (it, &item); switch (ires) { case GST_ITERATOR_DONE: done = TRUE; break; case GST_ITERATOR_OK: { GstPad *pad = g_value_get_object (&item); GstQuery *peerquery; GstClockTime min_cur, max_cur; gboolean live_cur; peerquery = gst_query_new_latency (); /* Ask peer for latency */ res &= gst_pad_peer_query (pad, peerquery); /* take max from all valid return values */ if (res) { gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur); if (min_cur > min) min = min_cur; if (max_cur != GST_CLOCK_TIME_NONE && ((max != GST_CLOCK_TIME_NONE && max_cur > max) || (max == GST_CLOCK_TIME_NONE))) max = max_cur; live = live || live_cur; } gst_query_unref (peerquery); g_value_reset (&item); break; } case GST_ITERATOR_RESYNC: live = FALSE; min = 0; max = GST_CLOCK_TIME_NONE; res = TRUE; gst_iterator_resync (it); break; default: res = FALSE; done = TRUE; break; } } g_value_unset (&item); gst_iterator_free (it); if (res) { /* store the results */ GST_DEBUG_OBJECT (audiomixer, "Calculated total latency: live %s, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, (live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max)); gst_query_set_latency (query, live, min, max); } return res; } static gboolean gst_audiomixer_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (parent); gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { GstFormat format; gst_query_parse_position (query, &format, NULL); switch (format) { case GST_FORMAT_TIME: /* FIXME, bring to stream time, might be tricky */ gst_query_set_position (query, format, audiomixer->segment.position); res = TRUE; break; case GST_FORMAT_DEFAULT: gst_query_set_position (query, format, audiomixer->offset); res = TRUE; break; default: break; } break; } case GST_QUERY_DURATION: res = gst_audiomixer_query_duration (audiomixer, query); break; case GST_QUERY_LATENCY: res = gst_audiomixer_query_latency (audiomixer, query); break; default: /* FIXME, needs a custom query handler because we have multiple * sinkpads */ res = gst_pad_query_default (pad, parent, query); break; } return res; } /* event handling */ typedef struct { GstEvent *event; gboolean flush; } EventData; /* FIXME: What is this supposed to solve? */ static gboolean forward_event_func (const GValue * val, GValue * ret, EventData * data) { GstPad *pad = g_value_get_object (val); GstEvent *event = data->event; GstPad *peer; gst_event_ref (event); GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event)); peer = gst_pad_get_peer (pad); /* collect pad might have been set flushing, * so bypass core checking that and send directly to peer */ if (!peer || !gst_pad_send_event (peer, event)) { if (!peer) gst_event_unref (event); GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.", event, GST_EVENT_TYPE_NAME (event)); /* quick hack to unflush the pads, ideally we need a way to just unflush * this single collect pad */ if (data->flush) gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE)); } else { g_value_set_boolean (ret, TRUE); GST_LOG_OBJECT (pad, "Sent event %p (%s).", event, GST_EVENT_TYPE_NAME (event)); } if (peer) gst_object_unref (peer); /* continue on other pads, even if one failed */ return TRUE; } /* forwards the event to all sinkpads, takes ownership of the * event * * Returns: TRUE if the event could be forwarded on all * sinkpads. */ static gboolean forward_event (GstAudioMixer * audiomixer, GstEvent * event, gboolean flush) { gboolean ret; GstIterator *it; GstIteratorResult ires; GValue vret = { 0 }; EventData data; GST_LOG_OBJECT (audiomixer, "Forwarding event %p (%s)", event, GST_EVENT_TYPE_NAME (event)); data.event = event; data.flush = flush; g_value_init (&vret, G_TYPE_BOOLEAN); g_value_set_boolean (&vret, FALSE); it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer)); while (TRUE) { ires = gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func, &vret, &data); switch (ires) { case GST_ITERATOR_RESYNC: GST_WARNING ("resync"); gst_iterator_resync (it); g_value_set_boolean (&vret, TRUE); break; case GST_ITERATOR_OK: case GST_ITERATOR_DONE: ret = g_value_get_boolean (&vret); goto done; default: ret = FALSE; goto done; } } done: gst_iterator_free (it); GST_LOG_OBJECT (audiomixer, "Forwarded event %p (%s), ret=%d", event, GST_EVENT_TYPE_NAME (event), ret); gst_event_unref (event); return ret; } static gboolean gst_audiomixer_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAudioMixer *audiomixer; gboolean result; audiomixer = GST_AUDIO_MIXER (parent); GST_DEBUG_OBJECT (pad, "Got %s event on src pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { /* TODO: Update from videomixer */ case GST_EVENT_SEEK: { GstSeekFlags flags; gdouble rate; GstSeekType start_type, stop_type; gint64 start, stop; GstFormat seek_format, dest_format; gboolean flush; /* parse the seek parameters */ gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); if ((start_type != GST_SEEK_TYPE_NONE) && (start_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (audiomixer, "seeking failed, unhandled seek type for start: %d", start_type); goto done; } if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (audiomixer, "seeking failed, unhandled seek type for end: %d", stop_type); goto done; } dest_format = audiomixer->segment.format; if (seek_format != dest_format) { result = FALSE; GST_DEBUG_OBJECT (audiomixer, "seeking failed, unhandled seek format: %d", seek_format); goto done; } flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH; /* check if we are flushing */ if (flush) { /* flushing seek, start flush downstream, the flush will be done * when all pads received a FLUSH_STOP. * Make sure we accept nothing anymore and return WRONG_STATE. * We send a flush-start before, to ensure no streaming is done * as we need to take the stream lock. */ gst_pad_push_event (audiomixer->srcpad, gst_event_new_flush_start ()); gst_collect_pads_set_flushing (audiomixer->collect, TRUE); /* We can't send FLUSH_STOP here since upstream could start pushing data * after we unlock audiomixer->collect. * We set flush_stop_pending to TRUE instead and send FLUSH_STOP after * forwarding the seek upstream or from gst_audiomixer_collected, * whichever happens first. */ GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect); audiomixer->flush_stop_pending = TRUE; GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect); GST_DEBUG_OBJECT (audiomixer, "mark pending flush stop event"); } GST_DEBUG_OBJECT (audiomixer, "handling seek event: %" GST_PTR_FORMAT, event); /* now wait for the collected to be finished and mark a new * segment. After we have the lock, no collect function is running and no * new collect function will be called for as long as we're flushing. */ GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect); /* clip position and update our segment */ if (audiomixer->segment.stop != -1) { audiomixer->segment.position = audiomixer->segment.stop; } gst_segment_do_seek (&audiomixer->segment, rate, seek_format, flags, start_type, start, stop_type, stop, NULL); if (flush) { /* Yes, we need to call _set_flushing again *WHEN* the streaming threads * have stopped so that the cookie gets properly updated. */ gst_collect_pads_set_flushing (audiomixer->collect, TRUE); } GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect); GST_DEBUG_OBJECT (audiomixer, "forwarding seek event: %" GST_PTR_FORMAT, event); GST_DEBUG_OBJECT (audiomixer, "updated segment: %" GST_SEGMENT_FORMAT, &audiomixer->segment); /* we're forwarding seek to all upstream peers and wait for one to reply * with a newsegment-event before we send a newsegment-event downstream */ g_atomic_int_set (&audiomixer->segment_pending, TRUE); result = forward_event (audiomixer, event, flush); /* FIXME: We should use the seek segment and forward that downstream next time * not any upstream segment event */ if (!result) { /* seek failed. maybe source is a live source. */ GST_DEBUG_OBJECT (audiomixer, "seeking failed"); } if (g_atomic_int_compare_and_exchange (&audiomixer->flush_stop_pending, TRUE, FALSE)) { GST_DEBUG_OBJECT (audiomixer, "pending flush stop"); if (!gst_pad_push_event (audiomixer->srcpad, gst_event_new_flush_stop (TRUE))) { GST_WARNING_OBJECT (audiomixer, "Sending flush stop event failed"); } } break; } case GST_EVENT_QOS: /* QoS might be tricky */ result = FALSE; gst_event_unref (event); break; case GST_EVENT_NAVIGATION: /* navigation is rather pointless. */ result = FALSE; gst_event_unref (event); break; default: /* just forward the rest for now */ GST_DEBUG_OBJECT (audiomixer, "forward unhandled event: %s", GST_EVENT_TYPE_NAME (event)); result = forward_event (audiomixer, event, FALSE); break; } done: return result; } static gboolean gst_audiomixer_sink_event (GstCollectPads * pads, GstCollectData * pad, GstEvent * event, gpointer user_data) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data); GstAudioMixerCollect *adata = (GstAudioMixerCollect *) pad; gboolean res = TRUE, discard = FALSE; GST_DEBUG_OBJECT (pad->pad, "Got %s event on sink pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); res = gst_audiomixer_setcaps (audiomixer, pad->pad, caps); gst_event_unref (event); event = NULL; break; } /* FIXME: Who cares about flushes from upstream? We should * not forward them at all */ case GST_EVENT_FLUSH_START: /* ensure that we will send a flush stop */ GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect); audiomixer->flush_stop_pending = TRUE; res = gst_collect_pads_event_default (pads, pad, event, discard); event = NULL; GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect); break; case GST_EVENT_FLUSH_STOP: /* we received a flush-stop. We will only forward it when * flush_stop_pending is set, and we will unset it then. */ g_atomic_int_set (&audiomixer->segment_pending, TRUE); GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect); if (audiomixer->flush_stop_pending) { GST_DEBUG_OBJECT (pad->pad, "forwarding flush stop"); res = gst_collect_pads_event_default (pads, pad, event, discard); audiomixer->flush_stop_pending = FALSE; event = NULL; gst_buffer_replace (&audiomixer->current_buffer, NULL); audiomixer->discont_time = GST_CLOCK_TIME_NONE; } else { discard = TRUE; GST_DEBUG_OBJECT (pad->pad, "eating flush stop"); } GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect); /* Clear pending tags */ if (audiomixer->pending_events) { g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (audiomixer->pending_events); audiomixer->pending_events = NULL; } adata->position = adata->size = 0; adata->output_offset = adata->next_offset = -1; gst_buffer_replace (&adata->buffer, NULL); break; case GST_EVENT_TAG: /* collect tags here so we can push them out when we collect data */ audiomixer->pending_events = g_list_append (audiomixer->pending_events, event); event = NULL; break; case GST_EVENT_SEGMENT:{ const GstSegment *segment; gst_event_parse_segment (event, &segment); if (segment->rate != audiomixer->segment.rate) { GST_ERROR_OBJECT (pad->pad, "Got segment event with wrong rate %lf, expected %lf", segment->rate, audiomixer->segment.rate); res = FALSE; gst_event_unref (event); event = NULL; } else if (segment->rate < 0.0) { GST_ERROR_OBJECT (pad->pad, "Negative rates not supported yet"); res = FALSE; gst_event_unref (event); event = NULL; } discard = TRUE; break; } default: break; } if (G_LIKELY (event)) return gst_collect_pads_event_default (pads, pad, event, discard); else return res; } static void gst_audiomixer_class_init (GstAudioMixerClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_audiomixer_set_property; gobject_class->get_property = gst_audiomixer_get_property; gobject_class->dispose = gst_audiomixer_dispose; g_object_class_install_property (gobject_class, PROP_FILTER_CAPS, g_param_spec_boxed ("caps", "Target caps", "Set target format for mixing (NULL means ANY). " "Setting this property takes a reference to the supplied GstCaps " "object", GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD, g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold", "Timestamp alignment threshold in nanoseconds", 0, G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT, g_param_spec_uint64 ("discont-wait", "Discont Wait", "Window of time in nanoseconds to wait before " "creating a discontinuity", 0, G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BLOCKSIZE, g_param_spec_uint ("blocksize", "Block Size", "Output block size in number of samples", 0, G_MAXUINT, DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audiomixer_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audiomixer_sink_template)); gst_element_class_set_static_metadata (gstelement_class, "AudioMixer", "Generic/Audio", "Mixes multiple audio streams", "Sebastian Dröge "); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_audiomixer_change_state); } static void gst_audiomixer_init (GstAudioMixer * audiomixer) { GstPadTemplate *template; template = gst_static_pad_template_get (&gst_audiomixer_src_template); audiomixer->srcpad = gst_pad_new_from_template (template, "src"); gst_object_unref (template); gst_pad_set_query_function (audiomixer->srcpad, GST_DEBUG_FUNCPTR (gst_audiomixer_src_query)); gst_pad_set_event_function (audiomixer->srcpad, GST_DEBUG_FUNCPTR (gst_audiomixer_src_event)); GST_PAD_SET_PROXY_CAPS (audiomixer->srcpad); gst_element_add_pad (GST_ELEMENT (audiomixer), audiomixer->srcpad); audiomixer->current_caps = NULL; gst_audio_info_init (&audiomixer->info); audiomixer->padcount = 0; audiomixer->filter_caps = NULL; audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD; audiomixer->discont_wait = DEFAULT_DISCONT_WAIT; audiomixer->blocksize = DEFAULT_BLOCKSIZE; /* keep track of the sinkpads requested */ audiomixer->collect = gst_collect_pads_new (); gst_collect_pads_set_function (audiomixer->collect, GST_DEBUG_FUNCPTR (gst_audiomixer_collected), audiomixer); gst_collect_pads_set_clip_function (audiomixer->collect, GST_DEBUG_FUNCPTR (gst_audiomixer_do_clip), audiomixer); gst_collect_pads_set_event_function (audiomixer->collect, GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event), audiomixer); gst_collect_pads_set_query_function (audiomixer->collect, GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query), audiomixer); } static void gst_audiomixer_dispose (GObject * object) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); if (audiomixer->collect) { gst_object_unref (audiomixer->collect); audiomixer->collect = NULL; } gst_caps_replace (&audiomixer->filter_caps, NULL); gst_caps_replace (&audiomixer->current_caps, NULL); if (audiomixer->pending_events) { g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (audiomixer->pending_events); audiomixer->pending_events = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_audiomixer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); switch (prop_id) { case PROP_FILTER_CAPS:{ GstCaps *new_caps = NULL; GstCaps *old_caps; const GstCaps *new_caps_val = gst_value_get_caps (value); if (new_caps_val != NULL) { new_caps = (GstCaps *) new_caps_val; gst_caps_ref (new_caps); } GST_OBJECT_LOCK (audiomixer); old_caps = audiomixer->filter_caps; audiomixer->filter_caps = new_caps; GST_OBJECT_UNLOCK (audiomixer); if (old_caps) gst_caps_unref (old_caps); GST_DEBUG_OBJECT (audiomixer, "set new caps %" GST_PTR_FORMAT, new_caps); break; } case PROP_ALIGNMENT_THRESHOLD: audiomixer->alignment_threshold = g_value_get_uint64 (value); break; case PROP_DISCONT_WAIT: audiomixer->discont_wait = g_value_get_uint64 (value); break; case PROP_BLOCKSIZE: audiomixer->blocksize = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); switch (prop_id) { case PROP_FILTER_CAPS: GST_OBJECT_LOCK (audiomixer); gst_value_set_caps (value, audiomixer->filter_caps); GST_OBJECT_UNLOCK (audiomixer); break; case PROP_ALIGNMENT_THRESHOLD: g_value_set_uint64 (value, audiomixer->alignment_threshold); break; case PROP_DISCONT_WAIT: g_value_set_uint64 (value, audiomixer->discont_wait); break; case PROP_BLOCKSIZE: g_value_set_uint (value, audiomixer->blocksize); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void free_pad (GstCollectData * data) { GstAudioMixerCollect *adata = (GstAudioMixerCollect *) data; gst_buffer_replace (&adata->buffer, NULL); } static GstPad * gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * unused, const GstCaps * caps) { gchar *name; GstAudioMixer *audiomixer; GstPad *newpad; gint padcount; GstCollectData *cdata; GstAudioMixerCollect *adata; if (templ->direction != GST_PAD_SINK) goto not_sink; audiomixer = GST_AUDIO_MIXER (element); /* increment pad counter */ padcount = g_atomic_int_add (&audiomixer->padcount, 1); name = g_strdup_printf ("sink_%u", padcount); newpad = g_object_new (GST_TYPE_AUDIO_MIXER_PAD, "name", name, "direction", templ->direction, "template", templ, NULL); GST_DEBUG_OBJECT (audiomixer, "request new pad %s", name); g_free (name); cdata = gst_collect_pads_add_pad (audiomixer->collect, newpad, sizeof (GstAudioMixerCollect), free_pad, TRUE); adata = (GstAudioMixerCollect *) cdata; adata->buffer = NULL; adata->position = 0; adata->size = 0; adata->output_offset = -1; adata->next_offset = -1; /* takes ownership of the pad */ if (!gst_element_add_pad (GST_ELEMENT (audiomixer), newpad)) goto could_not_add; gst_child_proxy_child_added (GST_CHILD_PROXY (audiomixer), G_OBJECT (newpad), GST_OBJECT_NAME (newpad)); return newpad; /* errors */ not_sink: { g_warning ("gstaudiomixer: request new pad that is not a SINK pad\n"); return NULL; } could_not_add: { GST_DEBUG_OBJECT (audiomixer, "could not add pad"); gst_collect_pads_remove_pad (audiomixer->collect, newpad); gst_object_unref (newpad); return NULL; } } static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad) { GstAudioMixer *audiomixer; audiomixer = GST_AUDIO_MIXER (element); GST_DEBUG_OBJECT (audiomixer, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); gst_child_proxy_child_removed (GST_CHILD_PROXY (audiomixer), G_OBJECT (pad), GST_OBJECT_NAME (pad)); if (audiomixer->collect) gst_collect_pads_remove_pad (audiomixer->collect, pad); gst_element_remove_pad (element, pad); } static GstFlowReturn gst_audiomixer_do_clip (GstCollectPads * pads, GstCollectData * data, GstBuffer * buffer, GstBuffer ** out, gpointer user_data) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data); gint rate, bpf; rate = GST_AUDIO_INFO_RATE (&audiomixer->info); bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf); *out = buffer; return GST_FLOW_OK; } static gboolean gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads, GstCollectData * collect_data, GstAudioMixerCollect * adata, GstBuffer * inbuf) { GstClockTime start_time, end_time; gboolean discont = FALSE; guint64 start_offset, end_offset; GstClockTime timestamp, stream_time; gint rate, bpf; g_assert (adata->buffer == NULL); rate = GST_AUDIO_INFO_RATE (&audiomixer->info); bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); timestamp = GST_BUFFER_TIMESTAMP (inbuf); stream_time = gst_segment_to_stream_time (&collect_data->segment, GST_FORMAT_TIME, timestamp); /* sync object properties on stream time */ /* TODO: Ideally we would want to do that on every sample */ if (GST_CLOCK_TIME_IS_VALID (stream_time)) gst_object_sync_values (GST_OBJECT (collect_data->pad), stream_time); adata->position = 0; adata->size = gst_buffer_get_size (inbuf); start_time = GST_BUFFER_TIMESTAMP (inbuf); end_time = start_time + gst_util_uint64_scale_ceil (adata->size / bpf, GST_SECOND, rate); start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND); end_offset = start_offset + adata->size / bpf; if (GST_BUFFER_IS_DISCONT (inbuf) || GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC) || adata->next_offset == -1) { discont = TRUE; } else { guint64 diff, max_sample_diff; /* Check discont, based on audiobasesink */ if (start_offset <= adata->next_offset) diff = adata->next_offset - start_offset; else diff = start_offset - adata->next_offset; max_sample_diff = gst_util_uint64_scale_int (audiomixer->alignment_threshold, rate, GST_SECOND); /* Discont! */ if (G_UNLIKELY (diff >= max_sample_diff)) { if (audiomixer->discont_wait > 0) { if (audiomixer->discont_time == GST_CLOCK_TIME_NONE) { audiomixer->discont_time = start_time; } else if (start_time - audiomixer->discont_time >= audiomixer->discont_wait) { discont = TRUE; audiomixer->discont_time = GST_CLOCK_TIME_NONE; } } else { discont = TRUE; } } else if (G_UNLIKELY (audiomixer->discont_time != GST_CLOCK_TIME_NONE)) { /* we have had a discont, but are now back on track! */ audiomixer->discont_time = GST_CLOCK_TIME_NONE; } } if (discont) { /* Have discont, need resync */ if (adata->next_offset != -1) GST_INFO_OBJECT (collect_data->pad, "Have discont. Expected %" G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT, adata->next_offset, start_offset); adata->output_offset = -1; } else { audiomixer->discont_time = GST_CLOCK_TIME_NONE; } adata->next_offset = end_offset; if (adata->output_offset == -1) { GstClockTime start_running_time; GstClockTime end_running_time; guint64 start_running_time_offset; guint64 end_running_time_offset; start_running_time = gst_segment_to_running_time (&collect_data->segment, GST_FORMAT_TIME, start_time); end_running_time = gst_segment_to_running_time (&collect_data->segment, GST_FORMAT_TIME, end_time); start_running_time_offset = gst_util_uint64_scale (start_running_time, rate, GST_SECOND); end_running_time_offset = gst_util_uint64_scale (end_running_time, rate, GST_SECOND); if (end_running_time_offset < audiomixer->offset) { /* Before output segment, drop */ gst_buffer_unref (inbuf); adata->buffer = NULL; gst_buffer_unref (gst_collect_pads_pop (pads, collect_data)); adata->position = 0; adata->size = 0; adata->output_offset = -1; GST_DEBUG_OBJECT (collect_data->pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset); return FALSE; } if (start_running_time_offset < audiomixer->offset) { guint diff = (audiomixer->offset - start_running_time_offset) * bpf; adata->position += diff; adata->size -= diff; /* FIXME: This could only happen due to rounding errors */ if (adata->size == 0) { /* Empty buffer, drop */ gst_buffer_unref (inbuf); adata->buffer = NULL; gst_buffer_unref (gst_collect_pads_pop (pads, collect_data)); adata->position = 0; adata->size = 0; adata->output_offset = -1; GST_DEBUG_OBJECT (collect_data->pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset); return FALSE; } } adata->output_offset = MAX (start_running_time_offset, audiomixer->offset); GST_DEBUG_OBJECT (collect_data->pad, "Buffer resynced: Pad offset %" G_GUINT64_FORMAT ", current mixer offset %" G_GUINT64_FORMAT, adata->output_offset, audiomixer->offset); } GST_LOG_OBJECT (collect_data->pad, "Queued new buffer at offset %" G_GUINT64_FORMAT, adata->output_offset); adata->buffer = inbuf; return TRUE; } static void gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads, GstCollectData * collect_data, GstAudioMixerCollect * adata, GstMapInfo * outmap) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (adata->collect.pad); guint overlap; guint out_start; GstBuffer *inbuf; GstMapInfo inmap; gint bpf; bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); /* Overlap => mix */ if (audiomixer->offset < adata->output_offset) out_start = adata->output_offset - audiomixer->offset; else out_start = 0; overlap = adata->size / bpf - adata->position / bpf; if (overlap > audiomixer->blocksize - out_start) overlap = audiomixer->blocksize - out_start; inbuf = gst_collect_pads_peek (pads, collect_data); g_assert (inbuf != NULL && inbuf == adata->buffer); GST_OBJECT_LOCK (pad); if (pad->mute || pad->volume < G_MINDOUBLE) { GST_DEBUG_OBJECT (pad, "Skipping muted pad"); gst_buffer_unref (inbuf); adata->position += overlap * bpf; adata->output_offset += overlap; if (adata->position >= adata->size) { /* Buffer done, drop it */ gst_buffer_replace (&adata->buffer, NULL); gst_buffer_unref (gst_collect_pads_pop (pads, collect_data)); } GST_OBJECT_UNLOCK (pad); return; } if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) { /* skip gap buffer */ GST_LOG_OBJECT (pad, "skipping GAP buffer"); gst_buffer_unref (inbuf); adata->output_offset += adata->size / bpf; /* Buffer done, drop it */ gst_buffer_replace (&adata->buffer, NULL); gst_buffer_unref (gst_collect_pads_pop (pads, collect_data)); GST_OBJECT_UNLOCK (pad); return; } gst_buffer_map (inbuf, &inmap, GST_MAP_READ); GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u", overlap * bpf, out_start * bpf, adata->position); /* further buffers, need to add them */ if (pad->volume == 1.0) { switch (audiomixer->info.finfo->format) { case GST_AUDIO_FORMAT_U8: audiomixer_orc_add_u8 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S8: audiomixer_orc_add_s8 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_U16: audiomixer_orc_add_u16 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S16: audiomixer_orc_add_s16 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_U32: audiomixer_orc_add_u32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S32: audiomixer_orc_add_s32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_F32: audiomixer_orc_add_f32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_F64: audiomixer_orc_add_f64 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), overlap * audiomixer->info.channels); break; default: g_assert_not_reached (); break; } } else { switch (audiomixer->info.finfo->format) { case GST_AUDIO_FORMAT_U8: audiomixer_orc_add_volume_u8 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i8, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S8: audiomixer_orc_add_volume_s8 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i8, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_U16: audiomixer_orc_add_volume_u16 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i16, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S16: audiomixer_orc_add_volume_s16 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i16, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_U32: audiomixer_orc_add_volume_u32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i32, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_S32: audiomixer_orc_add_volume_s32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume_i32, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_F32: audiomixer_orc_add_volume_f32 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume, overlap * audiomixer->info.channels); break; case GST_AUDIO_FORMAT_F64: audiomixer_orc_add_volume_f64 ((gpointer) (outmap->data + out_start * bpf), (gpointer) (inmap.data + adata->position), pad->volume, overlap * audiomixer->info.channels); break; default: g_assert_not_reached (); break; } } gst_buffer_unmap (inbuf, &inmap); gst_buffer_unref (inbuf); adata->position += overlap * bpf; adata->output_offset += overlap; if (adata->position == adata->size) { /* Buffer done, drop it */ gst_buffer_replace (&adata->buffer, NULL); gst_buffer_unref (gst_collect_pads_pop (pads, collect_data)); GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next"); } GST_OBJECT_UNLOCK (pad); } static GstFlowReturn gst_audiomixer_collected (GstCollectPads * pads, gpointer user_data) { /* Get all pads that have data for us and store them in a * new list. * * Calculate the current output offset/timestamp and * offset_end/timestamp_end. Allocate a silence buffer * for this and store it. * * For all pads: * 1) Once per input buffer (cached) * 1) Check discont (flag and timestamp with tolerance) * 2) If discont or new, resync. That means: * 1) Drop all start data of the buffer that comes before * the current position/offset. * 2) Calculate the offset (output segment!) that the first * frame of the input buffer corresponds to. Base this on * the running time. * * 2) If the current pad's offset/offset_end overlaps with the output * offset/offset_end, mix it at the appropiate position in the output * buffer and advance the pad's position. Remember if this pad needs * a new buffer to advance behind the output offset_end. * * 3) If we had no pad with a buffer, go EOS. * * 4) If we had at least one pad that did not advance behind output * offset_end, let collected be called again for the current * output offset/offset_end. */ GstAudioMixer *audiomixer; GSList *collected; GstFlowReturn ret; GstBuffer *outbuf = NULL; GstMapInfo outmap; gint64 next_offset; gint64 next_timestamp; gint rate, bpf; gboolean dropped = FALSE; gboolean is_eos = TRUE; gboolean is_done = TRUE; audiomixer = GST_AUDIO_MIXER (user_data); /* this is fatal */ if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) goto not_negotiated; if (audiomixer->flush_stop_pending == TRUE) { GST_INFO_OBJECT (audiomixer->srcpad, "send pending flush stop event"); if (!gst_pad_push_event (audiomixer->srcpad, gst_event_new_flush_stop (TRUE))) { GST_WARNING_OBJECT (audiomixer->srcpad, "Sending flush stop event failed"); } audiomixer->flush_stop_pending = FALSE; gst_buffer_replace (&audiomixer->current_buffer, NULL); audiomixer->discont_time = GST_CLOCK_TIME_NONE; } if (audiomixer->send_stream_start) { gchar s_id[32]; GstEvent *event; GST_INFO_OBJECT (audiomixer->srcpad, "send pending stream start event"); /* FIXME: create id based on input ids, we can't use * gst_pad_create_stream_id() though as that only handles 0..1 sink-pad */ g_snprintf (s_id, sizeof (s_id), "audiomixer-%08x", g_random_int ()); event = gst_event_new_stream_start (s_id); gst_event_set_group_id (event, gst_util_group_id_next ()); if (!gst_pad_push_event (audiomixer->srcpad, event)) { GST_WARNING_OBJECT (audiomixer->srcpad, "Sending stream start event failed"); } audiomixer->send_stream_start = FALSE; } if (audiomixer->send_caps) { GstEvent *caps_event; caps_event = gst_event_new_caps (audiomixer->current_caps); GST_INFO_OBJECT (audiomixer->srcpad, "send pending caps event %" GST_PTR_FORMAT, caps_event); if (!gst_pad_push_event (audiomixer->srcpad, caps_event)) { GST_WARNING_OBJECT (audiomixer->srcpad, "Sending caps event failed"); } audiomixer->send_caps = FALSE; } rate = GST_AUDIO_INFO_RATE (&audiomixer->info); bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); if (g_atomic_int_compare_and_exchange (&audiomixer->segment_pending, TRUE, FALSE)) { GstEvent *event; /* * When seeking we set the start and stop positions as given in the seek * event. We also adjust offset & timestamp accordingly. * This basically ignores all newsegments sent by upstream. * * FIXME: We require that all inputs have the same rate currently * as we do no rate conversion! */ event = gst_event_new_segment (&audiomixer->segment); if (audiomixer->segment.rate > 0.0) { audiomixer->segment.position = audiomixer->segment.start; } else { audiomixer->segment.position = audiomixer->segment.stop; } audiomixer->offset = gst_util_uint64_scale (audiomixer->segment.position, rate, GST_SECOND); GST_INFO_OBJECT (audiomixer->srcpad, "sending pending new segment event %" GST_SEGMENT_FORMAT, &audiomixer->segment); if (event) { if (!gst_pad_push_event (audiomixer->srcpad, event)) { GST_WARNING_OBJECT (audiomixer->srcpad, "Sending new segment event failed"); } } else { GST_WARNING_OBJECT (audiomixer->srcpad, "Creating new segment event for " "start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed", audiomixer->segment.start, audiomixer->segment.stop); } } if (G_UNLIKELY (audiomixer->pending_events)) { GList *tmp = audiomixer->pending_events; while (tmp) { GstEvent *ev = (GstEvent *) tmp->data; gst_pad_push_event (audiomixer->srcpad, ev); tmp = g_list_next (tmp); } g_list_free (audiomixer->pending_events); audiomixer->pending_events = NULL; } /* for the next timestamp, use the sample counter, which will * never accumulate rounding errors */ /* FIXME: Reverse mixing does not work at all yet */ if (audiomixer->segment.rate > 0.0) { next_offset = audiomixer->offset + audiomixer->blocksize; } else { next_offset = audiomixer->offset - audiomixer->blocksize; } next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate); if (audiomixer->current_buffer) { outbuf = audiomixer->current_buffer; } else { outbuf = gst_buffer_new_and_alloc (audiomixer->blocksize * bpf); gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE); gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data, outmap.size); gst_buffer_unmap (outbuf, &outmap); audiomixer->current_buffer = outbuf; } GST_LOG_OBJECT (audiomixer, "Starting to mix %u samples for offset %" G_GUINT64_FORMAT " with timestamp %" GST_TIME_FORMAT, audiomixer->blocksize, audiomixer->offset, GST_TIME_ARGS (audiomixer->segment.position)); gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); for (collected = pads->data; collected; collected = collected->next) { GstCollectData *collect_data; GstAudioMixerCollect *adata; GstBuffer *inbuf; collect_data = (GstCollectData *) collected->data; adata = (GstAudioMixerCollect *) collect_data; inbuf = gst_collect_pads_peek (pads, collect_data); if (!inbuf) continue; /* New buffer? */ if (!adata->buffer || adata->buffer != inbuf) { /* Takes ownership of buffer */ if (!gst_audio_mixer_fill_buffer (audiomixer, pads, collect_data, adata, inbuf)) { dropped = TRUE; continue; } } else { gst_buffer_unref (inbuf); } if (!adata->buffer && !dropped && GST_COLLECT_PADS_STATE_IS_SET (&adata->collect, GST_COLLECT_PADS_STATE_EOS)) { GST_DEBUG_OBJECT (collect_data->pad, "Pad is in EOS state"); } else { is_eos = FALSE; } /* At this point adata->output_offset >= audiomixer->offset or we have no buffer anymore */ if (adata->output_offset >= audiomixer->offset && adata->output_offset < audiomixer->offset + audiomixer->blocksize && adata->buffer) { GST_LOG_OBJECT (collect_data->pad, "Mixing buffer for current offset"); gst_audio_mixer_mix_buffer (audiomixer, pads, collect_data, adata, &outmap); if (adata->output_offset >= next_offset) { GST_DEBUG_OBJECT (collect_data->pad, "Pad is after current offset: %" G_GUINT64_FORMAT " >= %" G_GUINT64_FORMAT, adata->output_offset, next_offset); } else { is_done = FALSE; } } } gst_buffer_unmap (outbuf, &outmap); if (dropped) { /* We dropped a buffer, retry */ GST_DEBUG_OBJECT (audiomixer, "A pad dropped a buffer, wait for the next one"); return GST_FLOW_OK; } if (!is_done && !is_eos) { /* Get more buffers */ GST_DEBUG_OBJECT (audiomixer, "We're not done yet for the current offset," " waiting for more data"); return GST_FLOW_OK; } if (is_eos) { gint64 max_offset = 0; gboolean empty_buffer = TRUE; GST_DEBUG_OBJECT (audiomixer, "We're EOS"); for (collected = pads->data; collected; collected = collected->next) { GstCollectData *collect_data; GstAudioMixerCollect *adata; collect_data = (GstCollectData *) collected->data; adata = (GstAudioMixerCollect *) collect_data; max_offset = MAX (max_offset, adata->output_offset); if (adata->output_offset > audiomixer->offset) empty_buffer = FALSE; } /* This means EOS or no pads at all */ if (empty_buffer) { gst_buffer_replace (&audiomixer->current_buffer, NULL); goto eos; } if (max_offset <= next_offset) { GST_DEBUG_OBJECT (audiomixer, "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %" G_GUINT64_FORMAT, max_offset, next_offset); next_offset = max_offset; gst_buffer_resize (outbuf, 0, (next_offset - audiomixer->offset) * bpf); next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate); } } /* set timestamps on the output buffer */ if (audiomixer->segment.rate > 0.0) { GST_BUFFER_TIMESTAMP (outbuf) = audiomixer->segment.position; GST_BUFFER_OFFSET (outbuf) = audiomixer->offset; GST_BUFFER_OFFSET_END (outbuf) = next_offset; GST_BUFFER_DURATION (outbuf) = next_timestamp - audiomixer->segment.position; } else { GST_BUFFER_TIMESTAMP (outbuf) = next_timestamp; GST_BUFFER_OFFSET (outbuf) = next_offset; GST_BUFFER_OFFSET_END (outbuf) = audiomixer->offset; GST_BUFFER_DURATION (outbuf) = audiomixer->segment.position - next_timestamp; } audiomixer->offset = next_offset; audiomixer->segment.position = next_timestamp; /* send it out */ GST_LOG_OBJECT (audiomixer, "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %" G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_BUFFER_OFFSET (outbuf)); ret = gst_pad_push (audiomixer->srcpad, outbuf); audiomixer->current_buffer = NULL; GST_LOG_OBJECT (audiomixer, "pushed outbuf, result = %s", gst_flow_get_name (ret)); if (ret == GST_FLOW_OK && is_eos) goto eos; return ret; /* ERRORS */ not_negotiated: { GST_ELEMENT_ERROR (audiomixer, STREAM, FORMAT, (NULL), ("Unknown data received, not negotiated")); return GST_FLOW_NOT_NEGOTIATED; } eos: { GST_DEBUG_OBJECT (audiomixer, "EOS"); gst_pad_push_event (audiomixer->srcpad, gst_event_new_eos ()); return GST_FLOW_EOS; } } static GstStateChangeReturn gst_audiomixer_change_state (GstElement * element, GstStateChange transition) { GstAudioMixer *audiomixer; GstStateChangeReturn ret; audiomixer = GST_AUDIO_MIXER (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: audiomixer->offset = 0; audiomixer->flush_stop_pending = FALSE; audiomixer->segment_pending = TRUE; audiomixer->send_stream_start = TRUE; audiomixer->send_caps = TRUE; gst_caps_replace (&audiomixer->current_caps, NULL); gst_segment_init (&audiomixer->segment, GST_FORMAT_TIME); gst_collect_pads_start (audiomixer->collect); audiomixer->discont_time = GST_CLOCK_TIME_NONE; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* need to unblock the collectpads before calling the * parent change_state so that streaming can finish */ gst_collect_pads_stop (audiomixer->collect); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_buffer_replace (&audiomixer->current_buffer, NULL); break; default: break; } return ret; } /* GstChildProxy implementation */ static GObject * gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy, guint index) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy); GObject *obj = NULL; GST_OBJECT_LOCK (audiomixer); obj = g_list_nth_data (GST_ELEMENT_CAST (audiomixer)->sinkpads, index); if (obj) gst_object_ref (obj); GST_OBJECT_UNLOCK (audiomixer); return obj; } static guint gst_audiomixer_child_proxy_get_children_count (GstChildProxy * child_proxy) { guint count = 0; GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy); GST_OBJECT_LOCK (audiomixer); count = GST_ELEMENT_CAST (audiomixer)->numsinkpads; GST_OBJECT_UNLOCK (audiomixer); GST_INFO_OBJECT (audiomixer, "Children Count: %d", count); return count; } static void gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data) { GstChildProxyInterface *iface = g_iface; GST_INFO ("intializing child proxy interface"); iface->get_child_by_index = gst_audiomixer_child_proxy_get_child_by_index; iface->get_children_count = gst_audiomixer_child_proxy_get_children_count; } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiomixer", 0, "audio mixing element"); if (!gst_element_register (plugin, "audiomixer", GST_RANK_NONE, GST_TYPE_AUDIO_MIXER)) return FALSE; return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, audiomixer, "Mixes multiple audio streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)