/* GStreamer * Copyright (C) 1999 Erik Walthinsen * Copyright (C) 2003,2004 David A. Schleef * * 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. */ /* Element-Checklist-Version: 5 */ /** * SECTION:element-audioresample * * * Audioresample resamples raw audio buffers to different sample rates using * a configurable windowing function to enhance quality. * Example launch line * * * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink * * Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. * * * * Last reviewed on 2006-03-02 (0.10.4) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include /*#define DEBUG_ENABLED */ #include "gstaudioresample.h" #include #include GST_DEBUG_CATEGORY (audioresample_debug); #define GST_CAT_DEFAULT audioresample_debug /* elementfactory information */ static const GstElementDetails gst_audioresample_details = GST_ELEMENT_DETAILS ("Audio scaler", "Filter/Converter/Audio", "Resample audio", "David Schleef "); #define DEFAULT_FILTERLEN 16 enum { PROP_0, PROP_FILTERLEN }; #define SUPPORTED_CAPS \ GST_STATIC_CAPS ( \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 16, " \ "depth = (int) 16, " \ "signed = (boolean) true;" \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 32, " \ "depth = (int) 32, " \ "signed = (boolean) true;" \ "audio/x-raw-float, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 32; " \ "audio/x-raw-float, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 64" \ ) static GstStaticPadTemplate gst_audioresample_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS); static GstStaticPadTemplate gst_audioresample_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); static void gst_audioresample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audioresample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* vmethods */ static gboolean audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size); static GstCaps *audioresample_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps); static gboolean audioresample_transform_size (GstBaseTransform * trans, GstPadDirection direction, GstCaps * incaps, guint insize, GstCaps * outcaps, guint * outsize); static gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps); static GstFlowReturn audioresample_pushthrough (GstAudioresample * audioresample); static GstFlowReturn audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf); static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event); static gboolean audioresample_start (GstBaseTransform * base); static gboolean audioresample_stop (GstBaseTransform * base); static gboolean audioresample_query (GstPad * pad, GstQuery * query); static const GstQueryType *audioresample_query_type (GstPad * pad); #define DEBUG_INIT(bla) \ GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element"); GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform, GST_TYPE_BASE_TRANSFORM, DEBUG_INIT); static void gst_audioresample_base_init (gpointer g_class) { GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audioresample_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_audioresample_sink_template)); gst_element_class_set_details (gstelement_class, &gst_audioresample_details); } static void gst_audioresample_class_init (GstAudioresampleClass * klass) { GObjectClass *gobject_class; gobject_class = (GObjectClass *) klass; gobject_class->set_property = gst_audioresample_set_property; gobject_class->get_property = gst_audioresample_get_property; g_object_class_install_property (gobject_class, PROP_FILTERLEN, g_param_spec_int ("filter_length", "filter_length", "filter_length", 0, G_MAXINT, DEFAULT_FILTERLEN, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); GST_BASE_TRANSFORM_CLASS (klass)->start = GST_DEBUG_FUNCPTR (audioresample_start); GST_BASE_TRANSFORM_CLASS (klass)->stop = GST_DEBUG_FUNCPTR (audioresample_stop); GST_BASE_TRANSFORM_CLASS (klass)->transform_size = GST_DEBUG_FUNCPTR (audioresample_transform_size); GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = GST_DEBUG_FUNCPTR (audioresample_get_unit_size); GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = GST_DEBUG_FUNCPTR (audioresample_transform_caps); GST_BASE_TRANSFORM_CLASS (klass)->set_caps = GST_DEBUG_FUNCPTR (audioresample_set_caps); GST_BASE_TRANSFORM_CLASS (klass)->transform = GST_DEBUG_FUNCPTR (audioresample_transform); GST_BASE_TRANSFORM_CLASS (klass)->event = GST_DEBUG_FUNCPTR (audioresample_event); GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE; } static void gst_audioresample_init (GstAudioresample * audioresample, GstAudioresampleClass * klass) { GstBaseTransform *trans; trans = GST_BASE_TRANSFORM (audioresample); /* buffer alloc passthrough is too impossible. FIXME, it * is trivial in the passthrough case. */ gst_pad_set_bufferalloc_function (trans->sinkpad, NULL); audioresample->filter_length = DEFAULT_FILTERLEN; audioresample->need_discont = FALSE; gst_pad_set_query_function (trans->srcpad, audioresample_query); gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type); } /* vmethods */ static gboolean audioresample_start (GstBaseTransform * base) { GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); audioresample->resample = resample_new (); audioresample->ts_offset = -1; audioresample->offset = -1; audioresample->next_ts = -1; resample_set_filter_length (audioresample->resample, audioresample->filter_length); return TRUE; } static gboolean audioresample_stop (GstBaseTransform * base) { GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); if (audioresample->resample) { resample_free (audioresample->resample); audioresample->resample = NULL; } gst_caps_replace (&audioresample->sinkcaps, NULL); gst_caps_replace (&audioresample->srccaps, NULL); return TRUE; } static gboolean audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size) { gint width, channels; GstStructure *structure; gboolean ret; g_assert (size); /* this works for both float and int */ structure = gst_caps_get_structure (caps, 0); ret = gst_structure_get_int (structure, "width", &width); ret &= gst_structure_get_int (structure, "channels", &channels); g_return_val_if_fail (ret, FALSE); *size = width * channels / 8; return TRUE; } static GstCaps * audioresample_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps) { GstCaps *res; GstStructure *structure; /* transform caps gives one single caps so we can just replace * the rate property with our range. */ res = gst_caps_copy (caps); structure = gst_caps_get_structure (res, 0); gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); return res; } static gboolean resample_set_state_from_caps (ResampleState * state, GstCaps * incaps, GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate) { GstStructure *structure; gboolean ret; gint myinrate, myoutrate; int mychannels; gint width, depth; ResampleFormat format; GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %" GST_PTR_FORMAT, incaps, outcaps); structure = gst_caps_get_structure (incaps, 0); /* get width */ ret = gst_structure_get_int (structure, "width", &width); if (!ret) goto no_width; /* figure out the format */ if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) { if (width == 32) format = RESAMPLE_FORMAT_F32; else if (width == 64) format = RESAMPLE_FORMAT_F64; else goto wrong_depth; } else { /* for int, depth and width must be the same */ ret = gst_structure_get_int (structure, "depth", &depth); if (!ret || width != depth) goto not_equal; if (width == 16) format = RESAMPLE_FORMAT_S16; else if (width == 32) format = RESAMPLE_FORMAT_S32; else goto wrong_depth; } ret = gst_structure_get_int (structure, "rate", &myinrate); ret &= gst_structure_get_int (structure, "channels", &mychannels); if (!ret) goto no_in_rate_channels; structure = gst_caps_get_structure (outcaps, 0); ret = gst_structure_get_int (structure, "rate", &myoutrate); if (!ret) goto no_out_rate; if (channels) *channels = mychannels; if (inrate) *inrate = myinrate; if (outrate) *outrate = myoutrate; resample_set_format (state, format); resample_set_n_channels (state, mychannels); resample_set_input_rate (state, myinrate); resample_set_output_rate (state, myoutrate); return TRUE; /* ERRORS */ no_width: { GST_DEBUG ("failed to get width from caps"); return FALSE; } not_equal: { GST_DEBUG ("width %d and depth %d must be the same", width, depth); return FALSE; } wrong_depth: { GST_DEBUG ("unknown depth %d found", depth); return FALSE; } no_in_rate_channels: { GST_DEBUG ("could not get input rate and channels"); return FALSE; } no_out_rate: { GST_DEBUG ("could not get output rate"); return FALSE; } } static gboolean audioresample_transform_size (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps, guint * othersize) { GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); ResampleState *state; GstCaps *srccaps, *sinkcaps; gboolean use_internal = FALSE; /* whether we use the internal state */ gboolean ret = TRUE; GST_LOG_OBJECT (base, "asked to transform size %d in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC"); if (direction == GST_PAD_SINK) { sinkcaps = caps; srccaps = othercaps; } else { sinkcaps = othercaps; srccaps = caps; } /* if the caps are the ones that _set_caps got called with; we can use * our own state; otherwise we'll have to create a state */ if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) && gst_caps_is_equal (srccaps, audioresample->srccaps)) { use_internal = TRUE; state = audioresample->resample; } else { GST_DEBUG_OBJECT (audioresample, "caps are not the set caps, creating state"); state = resample_new (); resample_set_filter_length (state, audioresample->filter_length); resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL); } if (direction == GST_PAD_SINK) { /* asked to convert size of an incoming buffer */ *othersize = resample_get_output_size_for_input (state, size); } else { /* asked to convert size of an outgoing buffer */ *othersize = resample_get_input_size_for_output (state, size); } g_assert (*othersize % state->sample_size == 0); /* we make room for one extra sample, given that the resampling filter * can output an extra one for non-integral i_rate/o_rate */ GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize); if (!use_internal) { resample_free (state); } return ret; } static gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { gboolean ret; gint inrate, outrate; int channels; GstAudioresample *audioresample = GST_AUDIORESAMPLE (base); GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %" GST_PTR_FORMAT, incaps, outcaps); ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps, &channels, &inrate, &outrate); g_return_val_if_fail (ret, FALSE); audioresample->channels = channels; GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels); audioresample->i_rate = inrate; GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate); audioresample->o_rate = outrate; GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate); /* save caps so we can short-circuit in the size_transform if the caps * are the same */ gst_caps_replace (&audioresample->sinkcaps, incaps); gst_caps_replace (&audioresample->srccaps, outcaps); return TRUE; } static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event) { GstAudioresample *audioresample; audioresample = GST_AUDIORESAMPLE (base); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: break; case GST_EVENT_FLUSH_STOP: resample_input_flush (audioresample->resample); audioresample->ts_offset = -1; audioresample->next_ts = -1; audioresample->offset = -1; break; case GST_EVENT_NEWSEGMENT: resample_input_pushthrough (audioresample->resample); audioresample_pushthrough (audioresample); audioresample->ts_offset = -1; audioresample->next_ts = -1; audioresample->offset = -1; break; case GST_EVENT_EOS: resample_input_eos (audioresample->resample); audioresample_pushthrough (audioresample); break; default: break; } parent_class->event (base, event); return TRUE; } static GstFlowReturn audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf) { int outsize; int outsamples; ResampleState *r; r = audioresample->resample; outsize = resample_get_output_size (r); GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize); /* protect against mem corruption */ if (outsize > GST_BUFFER_SIZE (outbuf)) { GST_WARNING_OBJECT (audioresample, "overriding audioresample's outsize %d with outbuffer's size %d", outsize, GST_BUFFER_SIZE (outbuf)); outsize = GST_BUFFER_SIZE (outbuf); } /* catch possibly wrong size differences */ if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) { GST_WARNING_OBJECT (audioresample, "audioresample's outsize %d too far from outbuffer's size %d", outsize, GST_BUFFER_SIZE (outbuf)); } outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize); outsamples = outsize / r->sample_size; GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples", outsize, outsamples); GST_BUFFER_OFFSET (outbuf) = audioresample->offset; GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts; if (audioresample->ts_offset != -1) { audioresample->offset += outsamples; audioresample->ts_offset += outsamples; audioresample->next_ts = gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND, audioresample->o_rate); GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset; /* we calculate DURATION as the difference between "next" timestamp * and current timestamp so we ensure a contiguous stream, instead of * having rounding errors. */ GST_BUFFER_DURATION (outbuf) = audioresample->next_ts - GST_BUFFER_TIMESTAMP (outbuf); } else { /* no valid offset know, we can still sortof calculate the duration though */ GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale_int (outsamples, GST_SECOND, audioresample->o_rate); } /* check for possible mem corruption */ if (outsize > GST_BUFFER_SIZE (outbuf)) { /* this is an error that when it happens, would need fixing in the * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf), * and it gave us more ! */ GST_WARNING_OBJECT (audioresample, "audioresample, you memory corrupting bastard. " "you gave me outsize %d while my buffer was size %d", outsize, GST_BUFFER_SIZE (outbuf)); return GST_FLOW_ERROR; } /* catch possibly wrong size differences */ if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) { GST_WARNING_OBJECT (audioresample, "audioresample's written outsize %d too far from outbuffer's size %d", outsize, GST_BUFFER_SIZE (outbuf)); } GST_BUFFER_SIZE (outbuf) = outsize; if (G_UNLIKELY (audioresample->need_discont)) { GST_DEBUG_OBJECT (audioresample, "marking this buffer with the DISCONT flag"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); audioresample->need_discont = FALSE; } GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); return GST_FLOW_OK; } static gboolean audioresample_check_discont (GstAudioresample * audioresample, GstClockTime timestamp) { if (timestamp != GST_CLOCK_TIME_NONE && audioresample->prev_ts != GST_CLOCK_TIME_NONE && audioresample->prev_duration != GST_CLOCK_TIME_NONE && timestamp != audioresample->prev_ts + audioresample->prev_duration) { /* Potentially a discontinuous buffer. However, it turns out that many * elements generate imperfect streams due to rounding errors, so we permit * a small error (up to one sample) without triggering a filter * flush/restart (if triggered incorrectly, this will be audible) */ GstClockTimeDiff diff = timestamp - (audioresample->prev_ts + audioresample->prev_duration); if (ABS (diff) > GST_SECOND / audioresample->i_rate) { GST_WARNING_OBJECT (audioresample, "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff); return TRUE; } } return FALSE; } static GstFlowReturn audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf) { GstAudioresample *audioresample; ResampleState *r; guchar *data, *datacopy; gulong size; GstClockTime timestamp; audioresample = GST_AUDIORESAMPLE (base); r = audioresample->resample; data = GST_BUFFER_DATA (inbuf); size = GST_BUFFER_SIZE (inbuf); timestamp = GST_BUFFER_TIMESTAMP (inbuf); GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, size, GST_TIME_ARGS (timestamp), GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf)); /* check for timestamp discontinuities and flush/reset if needed */ if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) { /* Flush internal samples */ audioresample_pushthrough (audioresample); /* Inform downstream element about discontinuity */ audioresample->need_discont = TRUE; /* We want to recalculate the offset */ audioresample->ts_offset = -1; } if (audioresample->ts_offset == -1) { /* if we don't know the initial offset yet, calculate it based on the * input timestamp. */ if (GST_CLOCK_TIME_IS_VALID (timestamp)) { GstClockTime stime; /* offset used to calculate the timestamps. We use the sample offset for * this to make it more accurate. We want the first buffer to have the * same timestamp as the incoming timestamp. */ audioresample->next_ts = timestamp; audioresample->ts_offset = gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND); /* offset used to set as the buffer offset, this offset is always * relative to the stream time, note that timestamp is not... */ stime = (timestamp - base->segment.start) + base->segment.time; audioresample->offset = gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND); } } audioresample->prev_ts = timestamp; audioresample->prev_duration = GST_BUFFER_DURATION (inbuf); /* need to memdup, resample takes ownership. */ datacopy = g_memdup (data, size); resample_add_input_data (r, datacopy, size, g_free, datacopy); return audioresample_do_output (audioresample, outbuf); } /* push remaining data in the buffers out */ static GstFlowReturn audioresample_pushthrough (GstAudioresample * audioresample) { int outsize; ResampleState *r; GstBuffer *outbuf; GstFlowReturn res = GST_FLOW_OK; GstBaseTransform *trans; r = audioresample->resample; outsize = resample_get_output_size (r); if (outsize == 0) { GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush"); goto done; } trans = GST_BASE_TRANSFORM (audioresample); res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize, GST_PAD_CAPS (trans->srcpad), &outbuf); if (G_UNLIKELY (res != GST_FLOW_OK)) { GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes", outsize); goto done; } res = audioresample_do_output (audioresample, outbuf); if (G_UNLIKELY (res != GST_FLOW_OK)) goto done; res = gst_pad_push (trans->srcpad, outbuf); done: return res; } static gboolean audioresample_query (GstPad * pad, GstQuery * query) { GstAudioresample *audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad)); GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample); gboolean res = TRUE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { GstClockTime min, max; gboolean live; guint64 latency; GstPad *peer; gint rate = audioresample->i_rate; gint resampler_latency = audioresample->filter_length / 2; if (gst_base_transform_is_passthrough (trans)) resampler_latency = 0; if ((peer = gst_pad_get_peer (trans->sinkpad))) { if ((res = gst_pad_query (peer, query))) { gst_query_parse_latency (query, &live, &min, &max); GST_DEBUG ("Peer latency: min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); /* add our own latency */ if (rate != 0 && resampler_latency != 0) latency = gst_util_uint64_scale (resampler_latency, GST_SECOND, rate); else latency = 0; GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); min += latency; if (max != GST_CLOCK_TIME_NONE) max += latency; GST_DEBUG ("Calculated total latency : min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); gst_query_set_latency (query, live, min, max); } gst_object_unref (peer); } break; } default: res = gst_pad_query_default (pad, query); break; } gst_object_unref (audioresample); return res; } static const GstQueryType * audioresample_query_type (GstPad * pad) { static const GstQueryType types[] = { GST_QUERY_LATENCY, 0 }; return types; } static void gst_audioresample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioresample *audioresample; audioresample = GST_AUDIORESAMPLE (object); switch (prop_id) { case PROP_FILTERLEN: audioresample->filter_length = g_value_get_int (value); GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d", audioresample->filter_length); if (audioresample->resample) { resample_set_filter_length (audioresample->resample, audioresample->filter_length); gst_element_post_message (GST_ELEMENT (audioresample), gst_message_new_latency (GST_OBJECT (audioresample))); } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audioresample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioresample *audioresample; audioresample = GST_AUDIORESAMPLE (object); switch (prop_id) { case PROP_FILTERLEN: g_value_set_int (value, audioresample->filter_length); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean plugin_init (GstPlugin * plugin) { resample_init (); if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY, GST_TYPE_AUDIORESAMPLE)) { return FALSE; } return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "audioresample", "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);