/* GStreamer FAAC (Free AAC Encoder) plugin * Copyright (C) 2003 Ronald Bultje * * 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstfaac.h" #define SINK_CAPS \ "audio/x-raw-int, " \ "endianness = (int) BYTE_ORDER, " \ "signed = (boolean) true, " \ "width = (int) 16, " \ "depth = (int) 16, " \ "rate = (int) [ 8000, 96000 ], " \ "channels = (int) [ 1, 6 ] " /* these don't seem to work? */ #if 0 "audio/x-raw-int, " "endianness = (int) BYTE_ORDER, " "signed = (boolean) true, " "width = (int) 32, " "depth = (int) { 24, 32 }, " "rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]; " "audio/x-raw-float, " "endianness = (int) BYTE_ORDER, " "width = (int) 32, " "rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]" #endif #define SRC_CAPS \ "audio/mpeg, " \ "mpegversion = (int) { 4, 2 }, " \ "channels = (int) [ 1, 6 ], " \ "rate = (int) [ 8000, 96000 ]" static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (SRC_CAPS)); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (SINK_CAPS)); static const GstElementDetails gst_faac_details = GST_ELEMENT_DETAILS ("AAC audio encoder", "Codec/Encoder/Audio", "Free MPEG-2/4 AAC encoder", "Ronald Bultje "); enum { ARG_0, ARG_OUTPUTFORMAT, ARG_BITRATE, ARG_PROFILE, ARG_TNS, ARG_MIDSIDE, ARG_SHORTCTL }; static void gst_faac_base_init (GstFaacClass * klass); static void gst_faac_class_init (GstFaacClass * klass); static void gst_faac_init (GstFaac * faac); static void gst_faac_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_faac_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_faac_sink_event (GstPad * pad, GstEvent * event); static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps); static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * data); static GstStateChangeReturn gst_faac_change_state (GstElement * element, GstStateChange transition); static GstElementClass *parent_class = NULL; GST_DEBUG_CATEGORY_STATIC (faac_debug); #define GST_CAT_DEFAULT faac_debug #define FAAC_DEFAULT_MPEGVERSION 4 GType gst_faac_get_type (void) { static GType gst_faac_type = 0; if (!gst_faac_type) { static const GTypeInfo gst_faac_info = { sizeof (GstFaacClass), (GBaseInitFunc) gst_faac_base_init, NULL, (GClassInitFunc) gst_faac_class_init, NULL, NULL, sizeof (GstFaac), 0, (GInstanceInitFunc) gst_faac_init, }; const GInterfaceInfo preset_interface_info = { NULL, /* interface_init */ NULL, /* interface_finalize */ NULL /* interface_data */ }; gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT, "GstFaac", &gst_faac_info, 0); g_type_add_interface_static (gst_faac_type, GST_TYPE_PRESET, &preset_interface_info); } return gst_faac_type; } static void gst_faac_base_init (GstFaacClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); gst_element_class_set_details (element_class, &gst_faac_details); GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding"); } #define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ()) static GType gst_faac_profile_get_type (void) { static GType gst_faac_profile_type = 0; if (!gst_faac_profile_type) { static GEnumValue gst_faac_profile[] = { {MAIN, "MAIN", "Main profile"}, {LOW, "LC", "Low complexity profile"}, {SSR, "SSR", "Scalable sampling rate profile"}, {LTP, "LTP", "Long term prediction profile"}, {0, NULL, NULL}, }; gst_faac_profile_type = g_enum_register_static ("GstFaacProfile", gst_faac_profile); } return gst_faac_profile_type; } #define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ()) static GType gst_faac_shortctl_get_type (void) { static GType gst_faac_shortctl_type = 0; if (!gst_faac_shortctl_type) { static GEnumValue gst_faac_shortctl[] = { {SHORTCTL_NORMAL, "SHORTCTL_NORMAL", "Normal block type"}, {SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"}, {SHORTCTL_NOLONG, "SHORTCTL_NOLONG", "No long blocks"}, {0, NULL, NULL}, }; gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl", gst_faac_shortctl); } return gst_faac_shortctl_type; } #define GST_TYPE_FAAC_OUTPUTFORMAT (gst_faac_outputformat_get_type ()) static GType gst_faac_outputformat_get_type (void) { static GType gst_faac_outputformat_type = 0; if (!gst_faac_outputformat_type) { static GEnumValue gst_faac_outputformat[] = { {0, "OUTPUTFORMAT_RAW", "Raw AAC"}, {1, "OUTPUTFORMAT_ADTS", "ADTS headers"}, {0, NULL, NULL}, }; gst_faac_outputformat_type = g_enum_register_static ("GstFaacOutputFormat", gst_faac_outputformat); } return gst_faac_outputformat_type; } static void gst_faac_class_init (GstFaacClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); parent_class = g_type_class_peek_parent (klass); gobject_class->set_property = gst_faac_set_property; gobject_class->get_property = gst_faac_get_property; /* properties */ g_object_class_install_property (gobject_class, ARG_BITRATE, g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec", 8 * 1000, 320 * 1000, 128 * 1000, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_PROFILE, g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile", GST_TYPE_FAAC_PROFILE, MAIN, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_TNS, g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping", FALSE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_MIDSIDE, g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding", TRUE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_SHORTCTL, g_param_spec_enum ("shortctl", "Block type", "Block type encorcing", GST_TYPE_FAAC_SHORTCTL, MAIN, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_OUTPUTFORMAT, g_param_spec_enum ("outputformat", "Output format", "Format of output frames", GST_TYPE_FAAC_OUTPUTFORMAT, 0 /* RAW */ , G_PARAM_READWRITE)); /* virtual functions */ gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state); } static void gst_faac_init (GstFaac * faac) { faac->handle = NULL; faac->samplerate = -1; faac->channels = -1; faac->cache = NULL; faac->cache_time = GST_CLOCK_TIME_NONE; faac->cache_duration = 0; faac->next_ts = GST_CLOCK_TIME_NONE; faac->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_chain_function (faac->sinkpad, GST_DEBUG_FUNCPTR (gst_faac_chain)); gst_pad_set_setcaps_function (faac->sinkpad, GST_DEBUG_FUNCPTR (gst_faac_sink_setcaps)); gst_pad_set_event_function (faac->sinkpad, GST_DEBUG_FUNCPTR (gst_faac_sink_event)); gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad); faac->srcpad = gst_pad_new_from_static_template (&src_template, "src"); gst_pad_use_fixed_caps (faac->srcpad); gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad); /* default properties */ faac->bitrate = 1000 * 128; faac->profile = MAIN; faac->shortctl = SHORTCTL_NORMAL; faac->outputformat = 0; /* RAW */ faac->tns = FALSE; faac->midside = TRUE; } static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps) { GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad)); GstStructure *structure = gst_caps_get_structure (caps, 0); faacEncHandle *handle; gint channels, samplerate, width; gulong samples, bytes, fmt = 0, bps = 0; gboolean result = FALSE; if (!gst_caps_is_fixed (caps)) goto done; GST_OBJECT_LOCK (faac); if (faac->handle) { faacEncClose (faac->handle); faac->handle = NULL; } if (faac->cache) { gst_buffer_unref (faac->cache); faac->cache = NULL; } GST_OBJECT_UNLOCK (faac); if (!gst_structure_get_int (structure, "channels", &channels) || !gst_structure_get_int (structure, "rate", &samplerate)) { goto done; } if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes))) goto done; if (gst_structure_has_name (structure, "audio/x-raw-int")) { gst_structure_get_int (structure, "width", &width); switch (width) { case 16: fmt = FAAC_INPUT_16BIT; bps = 2; break; case 24: case 32: fmt = FAAC_INPUT_32BIT; bps = 4; break; default: g_return_val_if_reached (FALSE); } } else if (gst_structure_has_name (structure, "audio/x-raw-float")) { fmt = FAAC_INPUT_FLOAT; bps = 4; } if (!fmt) { faacEncClose (handle); goto done; } GST_OBJECT_LOCK (faac); faac->format = fmt; faac->bps = bps; faac->handle = handle; faac->bytes = bytes; faac->samples = samples; faac->channels = channels; faac->samplerate = samplerate; GST_OBJECT_UNLOCK (faac); result = TRUE; done: gst_object_unref (faac); return result; } static gboolean gst_faac_configure_source_pad (GstFaac * faac) { GstCaps *allowed_caps; GstCaps *srccaps; gboolean ret = FALSE; gint n, ver, mpegversion; faacEncConfiguration *conf; guint maxbitrate; mpegversion = FAAC_DEFAULT_MPEGVERSION; allowed_caps = gst_pad_get_allowed_caps (faac->srcpad); GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, allowed_caps); if (allowed_caps == NULL) return FALSE; if (gst_caps_is_empty (allowed_caps)) goto empty_caps; if (!gst_caps_is_any (allowed_caps)) { for (n = 0; n < gst_caps_get_size (allowed_caps); n++) { GstStructure *s = gst_caps_get_structure (allowed_caps, n); if (gst_structure_get_int (s, "mpegversion", &ver) && (ver == 4 || ver == 2)) { mpegversion = ver; break; } } } gst_caps_unref (allowed_caps); /* we negotiated caps update current configuration */ conf = faacEncGetCurrentConfiguration (faac->handle); conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2; conf->aacObjectType = faac->profile; conf->allowMidside = faac->midside; conf->useLfe = 0; conf->useTns = faac->tns; conf->bitRate = faac->bitrate / faac->channels; conf->inputFormat = faac->format; conf->outputFormat = faac->outputformat; conf->shortctl = faac->shortctl; /* check, warn and correct if the max bitrate for the given samplerate is * exceeded. Maximum of 6144 bit for a channel */ maxbitrate = (unsigned int) (6144.0 * (double) faac->samplerate / (double) 1024.0 + .5); if (conf->bitRate > maxbitrate) { GST_ELEMENT_WARNING (faac, RESOURCE, SETTINGS, (NULL), ("bitrate %lu exceeds maximum allowed bitrate of %u for samplerate %d. " "Setting bitrate to %u", conf->bitRate, maxbitrate, faac->samplerate, maxbitrate)); conf->bitRate = maxbitrate; } if (!faacEncSetConfiguration (faac->handle, conf)) goto set_failed; /* now create a caps for it all */ srccaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, mpegversion, "channels", G_TYPE_INT, faac->channels, "rate", G_TYPE_INT, faac->samplerate, NULL); if (mpegversion == 4) { GstBuffer *codec_data; guint8 *config = NULL; gulong config_len = 0; /* get the config string */ GST_DEBUG_OBJECT (faac, "retrieving decoder info"); faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len); /* copy it into a buffer */ codec_data = gst_buffer_new_and_alloc (config_len); memcpy (GST_BUFFER_DATA (codec_data), config, config_len); /* add to caps */ gst_caps_set_simple (srccaps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL); gst_buffer_unref (codec_data); } GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps); ret = gst_pad_set_caps (faac->srcpad, srccaps); gst_caps_unref (srccaps); return ret; /* ERROR */ empty_caps: { gst_caps_unref (allowed_caps); return FALSE; } set_failed: { GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration"); return FALSE; } } static gboolean gst_faac_sink_event (GstPad * pad, GstEvent * event) { GstFaac *faac; gboolean ret; faac = GST_FAAC (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: { GstBuffer *outbuf; if (!faac->handle) ret = FALSE; else ret = TRUE; /* flush first */ GST_DEBUG ("Pushing out remaining buffers because of EOS"); while (ret) { if (gst_pad_alloc_buffer_and_set_caps (faac->srcpad, GST_BUFFER_OFFSET_NONE, faac->bytes, GST_PAD_CAPS (faac->srcpad), &outbuf) == GST_FLOW_OK) { gint ret_size; GST_DEBUG ("next_ts %" GST_TIME_FORMAT, GST_TIME_ARGS (faac->next_ts)); if ((ret_size = faacEncEncode (faac->handle, NULL, 0, GST_BUFFER_DATA (outbuf), faac->bytes)) > 0) { GST_BUFFER_SIZE (outbuf) = ret_size; GST_BUFFER_TIMESTAMP (outbuf) = faac->next_ts; /* faac seems to always consume a fixed number of input samples, * therefore extrapolate the duration from that value and the incoming * bitrate */ GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (faac->samples, GST_SECOND, faac->channels * faac->samplerate); if (GST_CLOCK_TIME_IS_VALID (faac->next_ts)) faac->next_ts += GST_BUFFER_DURATION (outbuf); gst_pad_push (faac->srcpad, outbuf); } else { gst_buffer_unref (outbuf); ret = FALSE; } } else ret = FALSE; } ret = gst_pad_event_default (pad, event); break; } case GST_EVENT_NEWSEGMENT: ret = gst_pad_push_event (faac->srcpad, event); break; case GST_EVENT_TAG: ret = gst_pad_event_default (pad, event); break; default: ret = gst_pad_event_default (pad, event); break; } gst_object_unref (faac); return ret; } static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * inbuf) { GstFlowReturn result = GST_FLOW_OK; GstBuffer *outbuf, *subbuf; GstFaac *faac; guint size, ret_size, in_size, frame_size; faac = GST_FAAC (gst_pad_get_parent (pad)); if (!faac->handle) goto no_handle; if (!GST_PAD_CAPS (faac->srcpad)) { if (!gst_faac_configure_source_pad (faac)) goto nego_failed; } GST_DEBUG ("Got buffer time:%" GST_TIME_FORMAT " duration:%" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf))); size = GST_BUFFER_SIZE (inbuf); in_size = size; if (faac->cache) in_size += GST_BUFFER_SIZE (faac->cache); frame_size = faac->samples * faac->bps; while (1) { /* do we have enough data for one frame? */ if (in_size / faac->bps < faac->samples) { if (in_size > size) { GstBuffer *merge; /* this is panic! we got a buffer, but still don't have enough * data. Merge them and retry in the next cycle... */ merge = gst_buffer_merge (faac->cache, inbuf); gst_buffer_unref (faac->cache); gst_buffer_unref (inbuf); faac->cache = merge; } else if (in_size == size) { /* this shouldn't happen, but still... */ faac->cache = inbuf; } else if (in_size > 0) { faac->cache = gst_buffer_create_sub (inbuf, size - in_size, in_size); GST_BUFFER_DURATION (faac->cache) = GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (faac->cache) / size; GST_BUFFER_TIMESTAMP (faac->cache) = GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) * (size - in_size) / size); gst_buffer_unref (inbuf); } else { gst_buffer_unref (inbuf); } goto done; } /* create the frame */ if (in_size > size) { GstBuffer *merge; /* merge */ subbuf = gst_buffer_create_sub (inbuf, 0, frame_size - (in_size - size)); GST_BUFFER_DURATION (subbuf) = GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size; merge = gst_buffer_merge (faac->cache, subbuf); gst_buffer_unref (faac->cache); gst_buffer_unref (subbuf); subbuf = merge; faac->cache = NULL; } else { subbuf = gst_buffer_create_sub (inbuf, size - in_size, frame_size); GST_BUFFER_DURATION (subbuf) = GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size; GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) * (size - in_size) / size); } result = gst_pad_alloc_buffer_and_set_caps (faac->srcpad, GST_BUFFER_OFFSET_NONE, faac->bytes, GST_PAD_CAPS (faac->srcpad), &outbuf); if (result != GST_FLOW_OK) goto done; if ((ret_size = faacEncEncode (faac->handle, (gint32 *) GST_BUFFER_DATA (subbuf), GST_BUFFER_SIZE (subbuf) / faac->bps, GST_BUFFER_DATA (outbuf), faac->bytes)) < 0) { GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL)); gst_buffer_unref (inbuf); gst_buffer_unref (subbuf); result = GST_FLOW_ERROR; goto done; } if (ret_size > 0) { GST_BUFFER_SIZE (outbuf) = ret_size; if (faac->cache_time != GST_CLOCK_TIME_NONE) { GST_BUFFER_TIMESTAMP (outbuf) = faac->cache_time; faac->cache_time = GST_CLOCK_TIME_NONE; } else GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (subbuf); GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (subbuf); if (faac->cache_duration) { GST_BUFFER_DURATION (outbuf) += faac->cache_duration; faac->cache_duration = 0; } /* Store the value of the next expected timestamp to output * This is required in order to output the trailing encoded packets * at EOS with proper timestamps and duration. */ faac->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); GST_DEBUG ("Pushing out buffer time:%" GST_TIME_FORMAT " duration:%" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf))); result = gst_pad_push (faac->srcpad, outbuf); } else { /* FIXME: what I'm doing here isn't fully correct, but there * really isn't a better way yet. * Problem is that libfaac caches buffers (for encoding * purposes), so the timestamp of the outgoing buffer isn't * the same as the timestamp of the data that I pushed in. * However, I don't know the delay between those two so I * cannot really say aything about it. This is a bad guess. */ gst_buffer_unref (outbuf); if (faac->cache_time != GST_CLOCK_TIME_NONE) faac->cache_time = GST_BUFFER_TIMESTAMP (subbuf); faac->cache_duration += GST_BUFFER_DURATION (subbuf); } in_size -= frame_size; gst_buffer_unref (subbuf); } done: gst_object_unref (faac); return result; /* ERRORS */ no_handle: { GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL), ("format wasn't negotiated before chain function")); gst_buffer_unref (inbuf); result = GST_FLOW_ERROR; goto done; } nego_failed: { GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL), ("failed to negotiate MPEG/AAC format with next element")); gst_buffer_unref (inbuf); result = GST_FLOW_ERROR; goto done; } } static void gst_faac_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstFaac *faac = GST_FAAC (object); GST_OBJECT_LOCK (faac); switch (prop_id) { case ARG_BITRATE: faac->bitrate = g_value_get_int (value); break; case ARG_PROFILE: faac->profile = g_value_get_enum (value); break; case ARG_TNS: faac->tns = g_value_get_boolean (value); break; case ARG_MIDSIDE: faac->midside = g_value_get_boolean (value); break; case ARG_SHORTCTL: faac->shortctl = g_value_get_enum (value); break; case ARG_OUTPUTFORMAT: faac->outputformat = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (faac); } static void gst_faac_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstFaac *faac = GST_FAAC (object); GST_OBJECT_LOCK (faac); switch (prop_id) { case ARG_BITRATE: g_value_set_int (value, faac->bitrate); break; case ARG_PROFILE: g_value_set_enum (value, faac->profile); break; case ARG_TNS: g_value_set_boolean (value, faac->tns); break; case ARG_MIDSIDE: g_value_set_boolean (value, faac->midside); break; case ARG_SHORTCTL: g_value_set_enum (value, faac->shortctl); break; case ARG_OUTPUTFORMAT: g_value_set_enum (value, faac->outputformat); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (faac); } static GstStateChangeReturn gst_faac_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstFaac *faac = GST_FAAC (element); /* upwards state changes */ switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); /* downwards state changes */ switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: { GST_OBJECT_LOCK (faac); if (faac->handle) { faacEncClose (faac->handle); faac->handle = NULL; } if (faac->cache) { gst_buffer_unref (faac->cache); faac->cache = NULL; } faac->cache_time = GST_CLOCK_TIME_NONE; faac->cache_duration = 0; faac->samplerate = -1; faac->channels = -1; faac->next_ts = GST_CLOCK_TIME_NONE; GST_OBJECT_UNLOCK (faac); break; } default: break; } return ret; } static gboolean plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "faac", GST_RANK_SECONDARY, GST_TYPE_FAAC); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "faac", "Free AAC Encoder (FAAC)", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)