/* GStreamer FAAC (Free AAC Encoder) plugin * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> * Copyright (C) 2009 Mark Nauwelaerts <mnauw@users.sourceforge.net> * * 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-faac * @title: faac * @see_also: faad * * faac encodes raw audio to AAC (MPEG-4 part 3) streams. * * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv * ]| Encode a sine beep as aac and write to matroska container. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <stdlib.h> #include <string.h> #include <gst/audio/audio.h> #include <gst/pbutils/codec-utils.h> #include "gstfaac.h" #define SAMPLE_RATES " 8000, " \ "11025, " \ "12000, " \ "16000, " \ "22050, " \ "24000, " \ "32000, " \ "44100, " \ "48000, " \ "64000, " \ "88200, " \ "96000" /* 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, " \ "channels = (int) [ 1, 6 ], " \ "rate = (int) {" SAMPLE_RATES "}, " \ "stream-format = (string) { adts, raw }, " \ "base-profile = (string) { main, lc, ssr, ltp }, " \ "framed = (boolean) true; " \ "audio/mpeg, " \ "mpegversion = (int) 2, " \ "channels = (int) [ 1, 6 ], " \ "rate = (int) {" SAMPLE_RATES "}, " \ "stream-format = (string) { adts, raw }, " \ "profile = (string) { main, lc }," \ "framed = (boolean) true; " static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (SRC_CAPS)); enum { PROP_0, PROP_QUALITY, PROP_BITRATE, PROP_RATE_CONTROL, PROP_PROFILE, PROP_TNS, PROP_MIDSIDE, PROP_SHORTCTL }; enum { VBR = 1, ABR }; 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 GstCaps *gst_faac_enc_generate_sink_caps (void); static gboolean gst_faac_configure_source_pad (GstFaac * faac, GstAudioInfo * info); static gboolean gst_faac_stop (GstAudioEncoder * enc); static gboolean gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info); static GstFlowReturn gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf); GST_DEBUG_CATEGORY_STATIC (faac_debug); #define GST_CAT_DEFAULT faac_debug #define FAAC_DEFAULT_QUALITY 100 #define FAAC_DEFAULT_BITRATE 128 * 1000 #define FAAC_DEFAULT_RATE_CONTROL VBR #define FAAC_DEFAULT_TNS FALSE #define FAAC_DEFAULT_MIDSIDE TRUE #define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL #define gst_faac_parent_class parent_class G_DEFINE_TYPE (GstFaac, gst_faac, GST_TYPE_AUDIO_ENCODER); GST_ELEMENT_REGISTER_DEFINE (faac, "faac", GST_RANK_SECONDARY, GST_TYPE_FAAC); #define GST_TYPE_FAAC_RATE_CONTROL (gst_faac_brtype_get_type ()) static GType gst_faac_brtype_get_type (void) { static GType gst_faac_brtype_type = 0; if (!gst_faac_brtype_type) { static const GEnumValue gst_faac_brtype[] = { {VBR, "VBR", "VBR encoding"}, {ABR, "ABR", "ABR encoding"}, {0, NULL, NULL}, }; gst_faac_brtype_type = g_enum_register_static ("GstFaacBrtype", gst_faac_brtype); } return gst_faac_brtype_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 const 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; } static void gst_faac_class_init (GstFaacClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass); GstCaps *sink_caps; GstPadTemplate *sink_templ; gobject_class->set_property = gst_faac_set_property; gobject_class->get_property = gst_faac_get_property; GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding"); gst_element_class_add_static_pad_template (gstelement_class, &src_template); sink_caps = gst_faac_enc_generate_sink_caps (); sink_templ = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps); gst_element_class_add_pad_template (gstelement_class, sink_templ); gst_caps_unref (sink_caps); gst_element_class_set_static_metadata (gstelement_class, "AAC audio encoder", "Codec/Encoder/Audio", "Free MPEG-2/4 AAC encoder", "Ronald Bultje <rbultje@ronald.bitfreak.net>"); gst_type_mark_as_plugin_api (GST_TYPE_FAAC_RATE_CONTROL, 0); gst_type_mark_as_plugin_api (GST_TYPE_FAAC_SHORTCTL, 0); base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faac_handle_frame); /* properties */ g_object_class_install_property (gobject_class, PROP_QUALITY, g_param_spec_int ("quality", "Quality (%)", "Variable bitrate (VBR) quantizer quality in %", 1, 1000, FAAC_DEFAULT_QUALITY, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BITRATE, g_param_spec_int ("bitrate", "Bitrate (bps)", "Average Bitrate (ABR) in bits/sec", 8 * 1000, 320 * 1000, FAAC_DEFAULT_BITRATE, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RATE_CONTROL, g_param_spec_enum ("rate-control", "Rate Control (ABR/VBR)", "Encoding bitrate type (VBR/ABR)", GST_TYPE_FAAC_RATE_CONTROL, FAAC_DEFAULT_RATE_CONTROL, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_TNS, g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping", FAAC_DEFAULT_TNS, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MIDSIDE, g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding", FAAC_DEFAULT_MIDSIDE, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SHORTCTL, g_param_spec_enum ("shortctl", "Block type", "Block type encorcing", GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_faac_init (GstFaac * faac) { GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (faac)); } static void gst_faac_close_encoder (GstFaac * faac) { if (faac->handle) faacEncClose (faac->handle); faac->handle = NULL; } static gboolean gst_faac_stop (GstAudioEncoder * enc) { GstFaac *faac = GST_FAAC (enc); GST_DEBUG_OBJECT (faac, "stop"); gst_faac_close_encoder (faac); return TRUE; } static const GstAudioChannelPosition aac_channel_positions[][8] = { {GST_AUDIO_CHANNEL_POSITION_MONO}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, }, { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE1} }; static GstCaps * gst_faac_enc_generate_sink_caps (void) { GstCaps *caps = gst_caps_new_empty (); GstStructure *s, *t; gint i, c; static const int rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 }; GValue rates_arr = { 0, }; GValue tmp_v = { 0, }; g_value_init (&rates_arr, GST_TYPE_LIST); g_value_init (&tmp_v, G_TYPE_INT); for (i = 0; i < G_N_ELEMENTS (rates); i++) { g_value_set_int (&tmp_v, rates[i]); gst_value_list_append_value (&rates_arr, &tmp_v); } g_value_unset (&tmp_v); s = gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING, GST_AUDIO_NE (S16), "layout", G_TYPE_STRING, "interleaved", NULL); gst_structure_set_value (s, "rate", &rates_arr); t = gst_structure_copy (s); gst_structure_set (t, "channels", G_TYPE_INT, 1, NULL); gst_caps_append_structure (caps, t); for (i = 2; i <= 6; i++) { guint64 channel_mask = 0; t = gst_structure_copy (s); gst_structure_set (t, "channels", G_TYPE_INT, i, NULL); for (c = 0; c < i; c++) channel_mask |= G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c]; gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL); gst_caps_append_structure (caps, t); } gst_structure_free (s); g_value_unset (&rates_arr); GST_DEBUG ("Generated sinkcaps: %" GST_PTR_FORMAT, caps); return caps; } static void gst_faac_set_tags (GstFaac * faac) { GstTagList *taglist; /* create a taglist and add a bitrate tag to it */ taglist = gst_tag_list_new_empty (); gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, faac->bitrate, NULL); gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (faac), taglist, GST_TAG_MERGE_REPLACE); gst_tag_list_unref (taglist); } static gboolean gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info) { GstFaac *faac = GST_FAAC (enc); gint width; gulong fmt = 0; gboolean result = FALSE; /* base class takes care */ width = GST_AUDIO_INFO_WIDTH (info); if (GST_AUDIO_INFO_IS_INTEGER (info)) { switch (width) { case 16: fmt = FAAC_INPUT_16BIT; break; case 24: case 32: fmt = FAAC_INPUT_32BIT; break; default: g_return_val_if_reached (FALSE); } } else { fmt = FAAC_INPUT_FLOAT; } faac->format = fmt; /* finish up */ result = gst_faac_configure_source_pad (faac, info); if (!result) goto done; gst_faac_set_tags (faac); /* report needs to base class */ gst_audio_encoder_set_frame_samples_min (enc, faac->samples); gst_audio_encoder_set_frame_samples_max (enc, faac->samples); gst_audio_encoder_set_frame_max (enc, 1); done: return result; } /* check downstream caps to configure format */ static void gst_faac_negotiate (GstFaac * faac) { GstCaps *caps; /* default setup */ faac->profile = LOW; faac->mpegversion = 4; faac->outputformat = 0; caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (faac)); GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, caps); if (caps && gst_caps_get_size (caps) > 0) { GstStructure *s = gst_caps_get_structure (caps, 0); const gchar *str = NULL; gint i = 4; if ((str = gst_structure_get_string (s, "stream-format"))) { if (strcmp (str, "adts") == 0) { GST_DEBUG_OBJECT (faac, "use ADTS format for output"); faac->outputformat = 1; } else if (strcmp (str, "raw") == 0) { GST_DEBUG_OBJECT (faac, "use RAW format for output"); faac->outputformat = 0; } else { GST_DEBUG_OBJECT (faac, "unknown stream-format: %s", str); faac->outputformat = 0; } } if ((str = gst_structure_get_string (s, "profile"))) { if (strcmp (str, "main") == 0) { faac->profile = MAIN; } else if (strcmp (str, "lc") == 0) { faac->profile = LOW; } else if (strcmp (str, "ssr") == 0) { faac->profile = SSR; } else if (strcmp (str, "ltp") == 0) { faac->profile = LTP; } else { faac->profile = LOW; } } if (!gst_structure_get_int (s, "mpegversion", &i) || i == 4) { faac->mpegversion = 4; } else { faac->mpegversion = 2; } } if (caps) gst_caps_unref (caps); } static gboolean gst_faac_open_encoder (GstFaac * faac, GstAudioInfo * info) { faacEncHandle *handle; faacEncConfiguration *conf; guint maxbitrate; gulong samples, bytes; g_return_val_if_fail (info->rate != 0 && info->channels != 0, FALSE); /* clean up in case of re-configure */ gst_faac_close_encoder (faac); if (!(handle = faacEncOpen (info->rate, info->channels, &samples, &bytes))) goto setup_failed; /* mind channel count */ samples /= info->channels; /* record */ faac->handle = handle; faac->samples = samples; faac->bytes = bytes; GST_DEBUG_OBJECT (faac, "faac needs samples %d, output size %d", faac->samples, faac->bytes); /* we negotiated caps update current configuration */ conf = faacEncGetCurrentConfiguration (faac->handle); conf->mpegVersion = (faac->mpegversion == 4) ? MPEG4 : MPEG2; conf->aacObjectType = faac->profile; conf->allowMidside = faac->midside; conf->useLfe = 0; conf->useTns = faac->tns; if (faac->brtype == VBR) { conf->quantqual = faac->quality; } else if (faac->brtype == ABR) { conf->bitRate = faac->bitrate / info->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) info->rate / (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, info->rate, maxbitrate)); conf->bitRate = maxbitrate; } /* default 0 to start with, libfaac chooses based on bitrate */ conf->bandWidth = 0; if (!faacEncSetConfiguration (faac->handle, conf)) goto setup_failed; /* let's see what really happened, * note that this may not really match desired rate */ GST_DEBUG_OBJECT (faac, "average bitrate: %lu kbps", (conf->bitRate + 500) / 1000 * info->channels); GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual); GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth); return TRUE; /* ERRORS */ setup_failed: { GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL)); return FALSE; } } static gboolean gst_faac_configure_source_pad (GstFaac * faac, GstAudioInfo * info) { GstCaps *srccaps; gboolean ret; /* negotiate stream format */ gst_faac_negotiate (faac); if (!gst_faac_open_encoder (faac, info)) goto set_failed; /* now create a caps for it all */ srccaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, faac->mpegversion, "channels", G_TYPE_INT, info->channels, "rate", G_TYPE_INT, info->rate, "stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"), "framed", G_TYPE_BOOLEAN, TRUE, NULL); /* DecoderSpecificInfo is only available for mpegversion=4 */ if (faac->mpegversion == 4) { guint8 *config = NULL; gulong config_len = 0; /* get the config string */ GST_DEBUG_OBJECT (faac, "retrieving decoder info"); faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len); if (!gst_codec_utils_aac_caps_set_level_and_profile (srccaps, config, config_len)) { free (config); gst_caps_unref (srccaps); goto invalid_codec_data; } if (!faac->outputformat) { GstBuffer *codec_data; /* copy it into a buffer */ codec_data = gst_buffer_new_and_alloc (config_len); gst_buffer_fill (codec_data, 0, config, config_len); /* add to caps */ gst_caps_set_simple (srccaps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL); gst_buffer_unref (codec_data); } free (config); } else { const gchar *profile; /* Add least add the profile to the caps */ switch (faac->profile) { case MAIN: profile = "main"; break; case LTP: profile = "ltp"; break; case SSR: profile = "ssr"; break; case LOW: default: profile = "lc"; break; } gst_caps_set_simple (srccaps, "profile", G_TYPE_STRING, profile, NULL); /* FIXME: How to get the profile for mpegversion==2? */ } GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps); ret = gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (faac), srccaps); gst_caps_unref (srccaps); return ret; /* ERROR */ set_failed: { GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration"); return FALSE; } invalid_codec_data: { GST_ERROR_OBJECT (faac, "Invalid codec data"); return FALSE; } } static GstFlowReturn gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf) { GstFaac *faac = GST_FAAC (enc); GstFlowReturn ret = GST_FLOW_OK; GstBuffer *out_buf; gsize size, ret_size; int enc_ret; GstMapInfo map, omap; guint8 *data; GstAudioInfo *info = gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (faac)); out_buf = gst_buffer_new_and_alloc (faac->bytes); gst_buffer_map (out_buf, &omap, GST_MAP_WRITE); if (G_LIKELY (in_buf)) { if (memcmp (info->position, aac_channel_positions[info->channels - 1], sizeof (GstAudioChannelPosition) * info->channels) != 0) { in_buf = gst_buffer_make_writable (in_buf); gst_audio_buffer_reorder_channels (in_buf, info->finfo->format, info->channels, info->position, aac_channel_positions[info->channels - 1]); } gst_buffer_map (in_buf, &map, GST_MAP_READ); data = map.data; size = map.size; } else { data = NULL; size = 0; } if (G_UNLIKELY ((enc_ret = faacEncEncode (faac->handle, (gint32 *) data, size / (info->finfo->width / 8), omap.data, omap.size)) < 0)) goto encode_failed; ret_size = enc_ret; if (in_buf) gst_buffer_unmap (in_buf, &map); GST_LOG_OBJECT (faac, "encoder return: %" G_GSIZE_FORMAT, ret_size); if (ret_size > 0) { gst_buffer_unmap (out_buf, &omap); gst_buffer_resize (out_buf, 0, ret_size); ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples); } else { gst_buffer_unmap (out_buf, &omap); gst_buffer_unref (out_buf); /* re-create encoder after final flush */ if (!in_buf) { GST_DEBUG_OBJECT (faac, "flushed; recreating encoder"); gst_faac_close_encoder (faac); if (!gst_faac_open_encoder (faac, gst_audio_encoder_get_audio_info (enc))) ret = GST_FLOW_ERROR; } } return ret; /* ERRORS */ encode_failed: { GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL)); if (in_buf) gst_buffer_unmap (in_buf, &map); gst_buffer_unmap (out_buf, &omap); gst_buffer_unref (out_buf); return GST_FLOW_ERROR; } } 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 PROP_QUALITY: faac->quality = g_value_get_int (value); break; case PROP_BITRATE: faac->bitrate = g_value_get_int (value); break; case PROP_RATE_CONTROL: faac->brtype = g_value_get_enum (value); break; case PROP_TNS: faac->tns = g_value_get_boolean (value); break; case PROP_MIDSIDE: faac->midside = g_value_get_boolean (value); break; case PROP_SHORTCTL: faac->shortctl = 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 PROP_QUALITY: g_value_set_int (value, faac->quality); break; case PROP_BITRATE: g_value_set_int (value, faac->bitrate); break; case PROP_RATE_CONTROL: g_value_set_enum (value, faac->brtype); break; case PROP_TNS: g_value_set_boolean (value, faac->tns); break; case PROP_MIDSIDE: g_value_set_boolean (value, faac->midside); break; case PROP_SHORTCTL: g_value_set_enum (value, faac->shortctl); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (faac); } static gboolean plugin_init (GstPlugin * plugin) { return GST_ELEMENT_REGISTER (faac, plugin); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, faac, "Free AAC Encoder (FAAC)", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)