/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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-audiorate * @title: audiorate * @see_also: #GstVideoRate * * This element takes an incoming stream of timestamped raw audio frames and * produces a perfect stream by inserting or dropping samples as needed. * * This operation may be of use to link to elements that require or otherwise * implicitly assume a perfect stream as they do not store timestamps, * but derive this by some means (e.g. bitrate for some AVI cases). * * The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add * and #GstAudioRate:drop can be read to obtain information about number of * input samples, output samples, dropped samples (i.e. the number of unused * input samples) and inserted samples (i.e. the number of samples added to * stream). * * When the #GstAudioRate:silent property is set to FALSE, a GObject property * notification will be emitted whenever one of the #GstAudioRate:add or * #GstAudioRate:drop values changes. * This can potentially cause performance degradation. * Note that property notification will happen from the streaming thread, so * applications should be prepared for this. * * If the #GstAudioRate:tolerance property is non-zero, and an incoming buffer's * timestamp deviates less than the property indicates from what would make a * 'perfect time', then no samples will be added or dropped. * Note that the output is still guaranteed to be a perfect stream, which means * that the incoming data is then simply shifted (by less than the indicated * tolerance) to a perfect time. * * ## Example pipelines * |[ * gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav * ]| * Capture audio from the sound card and turn it into a perfect stream * for saving in a raw audio file. * |[ * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav * ]| * Decodes an audio file and transforms it into a perfect stream for saving * in a raw audio WAV file. Without the audio rate, the timing might not be * preserved correctly in the WAV file in case the decoded stream is jittery * or there are samples missing. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstaudiorate.h" #define GST_CAT_DEFAULT audio_rate_debug GST_DEBUG_CATEGORY_STATIC (audio_rate_debug); /* GstAudioRate signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; #define DEFAULT_SILENT TRUE #define DEFAULT_TOLERANCE (40 * GST_MSECOND) #define DEFAULT_SKIP_TO_FIRST FALSE enum { PROP_0, PROP_IN, PROP_OUT, PROP_ADD, PROP_DROP, PROP_SILENT, PROP_TOLERANCE, PROP_SKIP_TO_FIRST }; static GstStaticPadTemplate gst_audio_rate_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) ", layout = (string) { interleaved, non-interleaved }") ); static GstStaticPadTemplate gst_audio_rate_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) ", layout = (string) { interleaved, non-interleaved }") ); static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf); static void gst_audio_rate_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_rate_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element, GstStateChange transition); /*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */ static GParamSpec *pspec_drop = NULL; static GParamSpec *pspec_add = NULL; #define gst_audio_rate_parent_class parent_class G_DEFINE_TYPE (GstAudioRate, gst_audio_rate, GST_TYPE_ELEMENT); static void gst_audio_rate_class_init (GstAudioRateClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); object_class->set_property = gst_audio_rate_set_property; object_class->get_property = gst_audio_rate_get_property; g_object_class_install_property (object_class, PROP_IN, g_param_spec_uint64 ("in", "In", "Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (object_class, PROP_OUT, g_param_spec_uint64 ("out", "Out", "Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); pspec_add = g_param_spec_uint64 ("add", "Add", "Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS); g_object_class_install_property (object_class, PROP_ADD, pspec_add); pspec_drop = g_param_spec_uint64 ("drop", "Drop", "Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS); g_object_class_install_property (object_class, PROP_DROP, pspec_drop); g_object_class_install_property (object_class, PROP_SILENT, g_param_spec_boolean ("silent", "silent", "Don't emit notify for dropped and duplicated frames", DEFAULT_SILENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstAudioRate:tolerance: * * The difference between incoming timestamp and next timestamp must exceed * the given value for audiorate to add or drop samples. */ g_object_class_install_property (object_class, PROP_TOLERANCE, g_param_spec_uint64 ("tolerance", "tolerance", "Only act if timestamp jitter/imperfection exceeds indicated tolerance (ns)", 0, G_MAXUINT64, DEFAULT_TOLERANCE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstAudioRate:skip-to-first: * * Don't produce buffers before the first one we receive. */ g_object_class_install_property (object_class, PROP_SKIP_TO_FIRST, g_param_spec_boolean ("skip-to-first", "Skip to first buffer", "Don't produce buffers before the first one we receive", DEFAULT_SKIP_TO_FIRST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_set_static_metadata (element_class, "Audio rate adjuster", "Filter/Effect/Audio", "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream", "Wim Taymans "); gst_element_class_add_static_pad_template (element_class, &gst_audio_rate_sink_template); gst_element_class_add_static_pad_template (element_class, &gst_audio_rate_src_template); element_class->change_state = gst_audio_rate_change_state; } static void gst_audio_rate_reset (GstAudioRate * audiorate) { audiorate->next_offset = -1; audiorate->next_ts = -1; audiorate->discont = TRUE; gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED); gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME); GST_DEBUG_OBJECT (audiorate, "handle reset"); } static gboolean gst_audio_rate_setcaps (GstAudioRate * audiorate, GstCaps * caps) { GstAudioInfo info; if (!gst_audio_info_from_caps (&info, caps)) goto wrong_caps; audiorate->info = info; return TRUE; /* ERRORS */ wrong_caps: { GST_DEBUG_OBJECT (audiorate, "could not parse caps"); return FALSE; } } static void gst_audio_rate_init (GstAudioRate * audiorate) { audiorate->sinkpad = gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink"); gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event); gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain); GST_PAD_SET_PROXY_CAPS (audiorate->sinkpad); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad); audiorate->srcpad = gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src"); gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event); GST_PAD_SET_PROXY_CAPS (audiorate->srcpad); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad); audiorate->in = 0; audiorate->out = 0; audiorate->drop = 0; audiorate->add = 0; audiorate->silent = DEFAULT_SILENT; audiorate->tolerance = DEFAULT_TOLERANCE; } static void gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time) { GstBuffer *buf; GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT ", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts), GST_TIME_ARGS (time)); if (!GST_CLOCK_TIME_IS_VALID (time) || !GST_CLOCK_TIME_IS_VALID (audiorate->next_ts)) return; /* feed an empty buffer to chain with the given timestamp, * it will take care of filling */ buf = gst_buffer_new (); GST_BUFFER_TIMESTAMP (buf) = time; gst_audio_rate_chain (audiorate->sinkpad, GST_OBJECT_CAST (audiorate), buf); } static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean res; GstAudioRate *audiorate; audiorate = GST_AUDIO_RATE (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); if ((res = gst_audio_rate_setcaps (audiorate, caps))) { res = gst_pad_push_event (audiorate->srcpad, event); } else { gst_event_unref (event); } break; } case GST_EVENT_FLUSH_STOP: GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP"); gst_audio_rate_reset (audiorate); res = gst_pad_push_event (audiorate->srcpad, event); break; case GST_EVENT_SEGMENT: { gst_event_copy_segment (event, &audiorate->sink_segment); GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT"); #if 0 /* FIXME: bad things will likely happen if rate < 0 ... */ if (!update) { /* a new segment starts. We need to figure out what will be the next * sample offset. We mark the offsets as invalid so that the _chain * function will perform this calculation. */ gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop); #endif audiorate->next_offset = -1; audiorate->next_ts = -1; #if 0 } else { gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start); } #endif GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT, &audiorate->sink_segment); if (audiorate->sink_segment.format == GST_FORMAT_TIME) { /* TIME formats can be copied to src and forwarded */ res = gst_pad_push_event (audiorate->srcpad, event); gst_segment_copy_into (&audiorate->sink_segment, &audiorate->src_segment); } else { /* other formats will be handled in the _chain function */ gst_event_unref (event); res = TRUE; } break; } case GST_EVENT_EOS: /* Fill segment until the end */ if (GST_CLOCK_TIME_IS_VALID (audiorate->src_segment.stop)) gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop); res = gst_pad_push_event (audiorate->srcpad, event); break; case GST_EVENT_GAP: { /* Fill until end of gap */ GstClockTime timestamp, duration; gst_event_parse_gap (event, ×tamp, &duration); gst_event_unref (event); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { if (GST_CLOCK_TIME_IS_VALID (duration)) timestamp += duration; gst_audio_rate_fill_to_time (audiorate, timestamp); } res = TRUE; break; } default: res = gst_pad_event_default (pad, parent, event); break; } return res; } static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean res; GstAudioRate *audiorate; audiorate = GST_AUDIO_RATE (parent); switch (GST_EVENT_TYPE (event)) { default: res = gst_pad_push_event (audiorate->sinkpad, event); break; } return res; } static gboolean gst_audio_rate_convert (GstAudioRate * audiorate, GstFormat src_fmt, guint64 src_val, GstFormat dest_fmt, guint64 * dest_val) { return gst_audio_info_convert (&audiorate->info, src_fmt, src_val, dest_fmt, (gint64 *) dest_val); } static gboolean gst_audio_rate_convert_segments (GstAudioRate * audiorate) { GstFormat src_fmt, dst_fmt; src_fmt = audiorate->sink_segment.format; dst_fmt = audiorate->src_segment.format; #define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \ src_fmt, audiorate->sink_segment.field, \ dst_fmt, &audiorate->src_segment.field); audiorate->sink_segment.rate = audiorate->src_segment.rate; audiorate->sink_segment.flags = audiorate->src_segment.flags; audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate; CONVERT_VAL (start); CONVERT_VAL (stop); CONVERT_VAL (time); CONVERT_VAL (base); CONVERT_VAL (position); #undef CONVERT_VAL return TRUE; } static void gst_audio_rate_notify_drop (GstAudioRate * audiorate) { g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop); } static void gst_audio_rate_notify_add (GstAudioRate * audiorate) { g_object_notify_by_pspec ((GObject *) audiorate, pspec_add); } static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstAudioRate *audiorate; GstClockTime in_time; guint64 in_offset, in_offset_end, in_samples; guint in_size; GstFlowReturn ret = GST_FLOW_OK; GstClockTimeDiff diff; gint rate, bpf; GstAudioMeta *meta; audiorate = GST_AUDIO_RATE (parent); rate = GST_AUDIO_INFO_RATE (&audiorate->info); bpf = GST_AUDIO_INFO_BPF (&audiorate->info); /* need to be negotiated now */ if (bpf == 0) goto not_negotiated; /* we have a new pending segment */ if (audiorate->next_offset == -1) { gint64 pos; /* update the TIME segment */ gst_audio_rate_convert_segments (audiorate); /* first buffer, we are negotiated and we have a segment, calculate the * current expected offsets based on the segment.start, which is the first * media time of the segment and should match the media time of the first * buffer in that segment, which is the offset expressed in DEFAULT units. */ /* convert first timestamp of segment to sample position */ pos = gst_util_uint64_scale_int_round (audiorate->src_segment.start, GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos); /* resyncing is a discont */ audiorate->discont = TRUE; audiorate->next_offset = pos; audiorate->next_ts = gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND, GST_AUDIO_INFO_RATE (&audiorate->info)); if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead"); pos = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf), GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT, pos); audiorate->next_offset = pos; audiorate->next_ts = GST_BUFFER_TIMESTAMP (buf); } } in_time = GST_BUFFER_TIMESTAMP (buf); if (in_time == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time"); in_time = audiorate->next_ts; } meta = gst_buffer_get_audio_meta (buf); in_size = gst_buffer_get_size (buf); in_samples = meta ? meta->samples : in_size / bpf; audiorate->in += in_samples; /* calculate the buffer offset */ in_offset = gst_util_uint64_scale_int_round (in_time, rate, GST_SECOND); in_offset_end = in_offset + in_samples; GST_LOG_OBJECT (audiorate, "in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT ", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%" G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT ", ->next_ts:%" GST_TIME_FORMAT, GST_TIME_ARGS (in_time), GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, rate)), in_size, in_offset, in_offset_end, audiorate->next_offset, GST_TIME_ARGS (audiorate->next_ts)); diff = in_time - audiorate->next_ts; if (diff <= (GstClockTimeDiff) audiorate->tolerance && diff >= (GstClockTimeDiff) - audiorate->tolerance) { /* buffer time close enough to expected time, * so produce a perfect stream by simply 'shifting' * it to next ts and offset and sending */ GST_LOG_OBJECT (audiorate, "within tolerance %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->tolerance)); /* The outgoing buffer's offset will be set to ->next_offset, we also * need to adjust the offset_end value accordingly */ in_offset_end = audiorate->next_offset + in_samples; audiorate->out += in_samples; goto send; } /* do we need to insert samples */ if (in_offset > audiorate->next_offset) { GstBuffer *fill; gint fillsize; guint64 fillsamples; /* We don't want to allocate a single unreasonably huge buffer - it might be hundreds of megabytes. So, limit each output buffer to one second of audio */ fillsamples = in_offset - audiorate->next_offset; while (fillsamples > 0) { guint64 cursamples = MIN (fillsamples, rate); GstMapInfo fillmap; fillsamples -= cursamples; fillsize = cursamples * bpf; fill = gst_buffer_new_and_alloc (fillsize); gst_buffer_map (fill, &fillmap, GST_MAP_WRITE); gst_audio_format_fill_silence (audiorate->info.finfo, fillmap.data, fillmap.size); gst_buffer_unmap (fill, &fillmap); if (audiorate->info.layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) { gst_buffer_add_audio_meta (fill, &audiorate->info, cursamples, NULL); } GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples", cursamples); GST_BUFFER_OFFSET (fill) = audiorate->next_offset; audiorate->next_offset += cursamples; GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset; /* Use next timestamp, then calculate following timestamp based on * offset to get duration. Necessary complexity to get 'perfect' * streams */ GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts; audiorate->next_ts = gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND, rate); GST_BUFFER_DURATION (fill) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (fill); /* we created this buffer to fill a gap */ GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP); /* set discont if it's pending, this is mostly done for the first buffer * and after a flushing seek */ if (audiorate->discont) { GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT); audiorate->discont = FALSE; } fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf); if (fill) ret = gst_pad_push (audiorate->srcpad, fill); if (ret != GST_FLOW_OK) goto beach; audiorate->out += cursamples; audiorate->add += cursamples; if (!audiorate->silent) gst_audio_rate_notify_add (audiorate); } } else if (in_offset < audiorate->next_offset) { /* need to remove samples */ if (in_offset_end <= audiorate->next_offset) { guint64 drop = in_samples; audiorate->drop += drop; GST_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " samples", drop); /* we can drop the buffer completely */ gst_buffer_unref (buf); buf = NULL; if (!audiorate->silent) gst_audio_rate_notify_drop (audiorate); goto beach; } else { guint64 truncsamples, leftsamples; /* truncate buffer */ truncsamples = audiorate->next_offset - in_offset; leftsamples = in_samples - truncsamples; buf = gst_audio_buffer_truncate (buf, bpf, truncsamples, leftsamples); audiorate->drop += truncsamples; audiorate->out += leftsamples; GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples", truncsamples); if (!audiorate->silent) gst_audio_rate_notify_drop (audiorate); } } send: if (gst_buffer_get_size (buf) == 0) goto beach; /* Now calculate parameters for whichever buffer (either the original * or truncated one) we're pushing. */ GST_BUFFER_OFFSET (buf) = audiorate->next_offset; GST_BUFFER_OFFSET_END (buf) = in_offset_end; GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts; audiorate->next_ts = gst_util_uint64_scale_int_round (in_offset_end, GST_SECOND, rate); GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf); if (audiorate->discont) { /* we need to output a discont buffer, do so now */ GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer"); buf = gst_buffer_make_writable (buf); GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); audiorate->discont = FALSE; } else if (GST_BUFFER_IS_DISCONT (buf)) { /* else we make everything continuous so we can safely remove the DISCONT * flag from the buffer if there was one */ GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer"); buf = gst_buffer_make_writable (buf); GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT); } buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf); if (buf) { /* set last_stop on segment */ audiorate->src_segment.position = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); ret = gst_pad_push (audiorate->srcpad, buf); } buf = NULL; audiorate->next_offset = in_offset_end; beach: if (buf) gst_buffer_unref (buf); return ret; /* ERRORS */ not_negotiated: { gst_buffer_unref (buf); GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT, (NULL), ("pipeline error, format was not negotiated")); return GST_FLOW_NOT_NEGOTIATED; } } static void gst_audio_rate_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioRate *audiorate = GST_AUDIO_RATE (object); switch (prop_id) { case PROP_SILENT: audiorate->silent = g_value_get_boolean (value); break; case PROP_TOLERANCE: audiorate->tolerance = g_value_get_uint64 (value); break; case PROP_SKIP_TO_FIRST: audiorate->skip_to_first = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_rate_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioRate *audiorate = GST_AUDIO_RATE (object); switch (prop_id) { case PROP_IN: g_value_set_uint64 (value, audiorate->in); break; case PROP_OUT: g_value_set_uint64 (value, audiorate->out); break; case PROP_ADD: g_value_set_uint64 (value, audiorate->add); break; case PROP_DROP: g_value_set_uint64 (value, audiorate->drop); break; case PROP_SILENT: g_value_set_boolean (value, audiorate->silent); break; case PROP_TOLERANCE: g_value_set_uint64 (value, audiorate->tolerance); break; case PROP_SKIP_TO_FIRST: g_value_set_boolean (value, audiorate->skip_to_first); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element, GstStateChange transition) { GstAudioRate *audiorate = GST_AUDIO_RATE (element); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: audiorate->in = 0; audiorate->out = 0; audiorate->drop = 0; audiorate->add = 0; gst_audio_info_init (&audiorate->info); gst_audio_rate_reset (audiorate); break; default: break; } return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0, "AudioRate stream fixer"); return gst_element_register (plugin, "audiorate", GST_RANK_NONE, GST_TYPE_AUDIO_RATE); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, audiorate, "Adjusts audio frames", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)