/* 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 "gstfaac.h" GST_PAD_TEMPLATE_FACTORY (src_template, "src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_CAPS_NEW ( "faac_mpeg_templ", "audio/mpeg", "systemstream", GST_PROPS_BOOLEAN (FALSE), "mpegversion", GST_PROPS_LIST ( GST_PROPS_INT (4), /* we prefer 4 */ GST_PROPS_INT (2) ), "channels", GST_PROPS_INT_RANGE (1, 6), "samplerate", GST_PROPS_INT_RANGE (8000, 96000) ) ); GST_PAD_TEMPLATE_FACTORY (sink_template, "sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_CAPS_NEW ( "faac_int16_templ", "audio/x-raw-int", "endianness", GST_PROPS_INT (G_BYTE_ORDER), "signed", GST_PROPS_BOOLEAN (TRUE), "width", GST_PROPS_INT (16), "depth", GST_PROPS_INT (16), "rate", GST_PROPS_INT_RANGE (8000, 96000), "channels", GST_PROPS_INT_RANGE (1, 6) ), GST_CAPS_NEW ( "faac_int24_templ", "audio/x-raw-int", "endianness", GST_PROPS_INT (G_BYTE_ORDER), "signed", GST_PROPS_BOOLEAN (TRUE), "width", GST_PROPS_INT (32), "depth", GST_PROPS_INT (24), "rate", GST_PROPS_INT_RANGE (8000, 96000), "channels", GST_PROPS_INT_RANGE (1, 6) ), GST_CAPS_NEW ( "faac_float_templ", "audio/x-raw-float", "endianness", GST_PROPS_INT (G_BYTE_ORDER), "depth", GST_PROPS_INT (32), /* float */ "rate", GST_PROPS_INT_RANGE (8000, 96000), "channels", GST_PROPS_INT_RANGE (1, 6) ) ); enum { ARG_0, ARG_BITRATE, ARG_PROFILE, ARG_TNS, ARG_MIDSIDE, ARG_SHORTCTL /* FILL ME */ }; 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 GstPadLinkReturn gst_faac_sinkconnect (GstPad *pad, GstCaps *caps); static GstPadLinkReturn gst_faac_srcconnect (GstPad *pad, GstCaps *caps); static void gst_faac_chain (GstPad *pad, GstData *data); static GstElementStateReturn gst_faac_change_state (GstElement *element); static GstElementClass *parent_class = NULL; /* static guint gst_faac_signals[LAST_SIGNAL] = { 0 }; */ 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, }; gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT, "GstFaac", &gst_faac_info, 0); } return gst_faac_type; } static void gst_faac_base_init (GstFaacClass *klass) { GstElementDetails gst_faac_details = { "Free AAC Encoder (FAAC)", "Codec/Audio/Encoder", "Free MPEG-2/4 AAC encoder", "Ronald Bultje ", }; GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, GST_PAD_TEMPLATE_GET (src_template)); gst_element_class_add_pad_template (element_class, GST_PAD_TEMPLATE_GET (sink_template)); gst_element_class_set_details (element_class, &gst_faac_details); } #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, "LOW", "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; } 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_ref (GST_TYPE_ELEMENT); /* properties */ g_object_class_install_property (gobject_class, ARG_BITRATE, g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec", 8 * 1024, 320 * 1024, 128 * 1024, 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)); /* virtual functions */ gstelement_class->change_state = gst_faac_change_state; gobject_class->set_property = gst_faac_set_property; gobject_class->get_property = gst_faac_get_property; } 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; GST_FLAG_SET (faac, GST_ELEMENT_EVENT_AWARE); faac->sinkpad = gst_pad_new_from_template ( GST_PAD_TEMPLATE_GET (sink_template), "sink"); gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad); gst_pad_set_chain_function (faac->sinkpad, gst_faac_chain); gst_pad_set_link_function (faac->sinkpad, gst_faac_sinkconnect); faac->srcpad = gst_pad_new_from_template ( GST_PAD_TEMPLATE_GET (src_template), "src"); gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad); gst_pad_set_link_function (faac->srcpad, gst_faac_srcconnect); /* default properties */ faac->bitrate = 1024 * 128; faac->profile = MAIN; faac->shortctl = SHORTCTL_NORMAL; faac->tns = FALSE; faac->midside = TRUE; } static GstPadLinkReturn gst_faac_sinkconnect (GstPad *pad, GstCaps *caps) { GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad)); if (!GST_CAPS_IS_FIXED (caps)) return GST_PAD_LINK_DELAYED; if (faac->handle) { faacEncClose (faac->handle); faac->handle = NULL; } if (faac->cache) { gst_buffer_unref (faac->cache); faac->cache = NULL; } for (; caps != NULL; caps = caps->next) { faacEncHandle *handle; gint channels, samplerate, depth; gulong samples, bytes, fmt = 0, bps = 0; gst_caps_get (caps, "channels", &channels, "rate", &samplerate, "depth", &depth, NULL); /* open a new handle to the encoder */ if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes))) continue; switch (depth) { case 16: fmt = FAAC_INPUT_16BIT; bps = 2; break; case 24: fmt = FAAC_INPUT_32BIT; /* 24-in-32, actually */ bps = 4; break; case 32: fmt = FAAC_INPUT_FLOAT; /* see template, this is right */ bps = 4; break; } if (!fmt) { faacEncClose (handle); continue; } faac->format = fmt; faac->bps = bps; faac->handle = handle; faac->bytes = bytes; faac->samples = samples; faac->channels = channels; faac->samplerate = samplerate; /* if the other side was already set-up, redo that */ if (GST_PAD_CAPS (faac->srcpad)) return gst_faac_srcconnect (faac->srcpad, gst_pad_get_allowed_caps (faac->srcpad)); /* else, that'll be done later */ return GST_PAD_LINK_OK; } return GST_PAD_LINK_REFUSED; } static GstPadLinkReturn gst_faac_srcconnect (GstPad *pad, GstCaps *caps) { GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad)); GstCaps *t; if (!faac->handle || (faac->samplerate == -1 || faac->channels == -1)) { return GST_PAD_LINK_DELAYED; } /* we do samplerate/channels ourselves */ for (t = caps; t != NULL; t = t->next) { gst_props_remove_entry_by_name (t->properties, "rate"); gst_props_remove_entry_by_name (t->properties, "channels"); } /* go through list */ caps = gst_caps_normalize (caps); for ( ; caps != NULL; caps = caps->next) { faacEncConfiguration *conf; gint mpegversion = 0; GstCaps *newcaps; GstPadLinkReturn ret; gst_caps_get_int (caps, "mpegversion", &mpegversion); /* new conf */ 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; conf->inputFormat = faac->format; /* FIXME: this one here means that we do not support direct * "MPEG audio file" output (like mp3). This means we can * only mux this into mov/qt (mp4a) or matroska or so. If * we want to support direct AAC file output, we need ADTS * headers, and we need to find a way in the caps to detect * that (that the next element is filesink or any element * that does want ADTS headers). */ conf->outputFormat = 0; /* raw, no ADTS headers */ conf->shortctl = faac->shortctl; if (!faacEncSetConfiguration (faac->handle, conf)) { GST_WARNING ("Faac doesn't support the current conf"); continue; } newcaps = GST_CAPS_NEW ("faac_mpeg_caps", "audio/mpeg", "systemstream", GST_PROPS_BOOLEAN (FALSE), "mpegversion", GST_PROPS_INT (mpegversion), "channels", GST_PROPS_INT (faac->channels), "rate", GST_PROPS_INT (faac->samplerate)); ret = gst_pad_try_set_caps (faac->srcpad, newcaps); switch (ret) { case GST_PAD_LINK_OK: case GST_PAD_LINK_DONE: return GST_PAD_LINK_DONE; case GST_PAD_LINK_DELAYED: return GST_PAD_LINK_DELAYED; default: break; } } return GST_PAD_LINK_REFUSED; } static void gst_faac_chain (GstPad *pad, GstData *data) { GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad)); GstBuffer *inbuf, *outbuf, *subbuf; guint size, ret_size, in_size, frame_size; if (GST_IS_EVENT (data)) { GstEvent *event = GST_EVENT (data); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* flush first */ while (1) { outbuf = gst_buffer_new_and_alloc (faac->bytes); if ((ret_size = faacEncEncode (faac->handle, NULL, 0, GST_BUFFER_DATA (outbuf), faac->bytes)) < 0) { gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding"); gst_event_unref (event); gst_buffer_unref (outbuf); return; } if (ret_size > 0) { GST_BUFFER_SIZE (outbuf) = ret_size; GST_BUFFER_TIMESTAMP (outbuf) = 0; GST_BUFFER_DURATION (outbuf) = 0; gst_pad_push (faac->srcpad, GST_DATA (outbuf)); } else { break; } } gst_element_set_eos (GST_ELEMENT (faac)); gst_pad_push (faac->srcpad, data); return; default: gst_pad_event_default (pad, event); return; } } inbuf = GST_BUFFER (data); if (!faac->handle) { gst_element_error (GST_ELEMENT (faac), "No input format negotiated"); gst_buffer_unref (inbuf); return; } if (!GST_PAD_CAPS (faac->srcpad)) { if (gst_faac_srcconnect (faac->srcpad, gst_pad_get_allowed_caps (faac->srcpad)) <= 0) { gst_element_error (GST_ELEMENT (faac), "Failed to negotiate MPEG/AAC format with next element"); gst_buffer_unref (inbuf); return; } } 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) { /* this is panic! we got a buffer, but still don't have enough * data. Merge them and retry in the next cycle... */ faac->cache = gst_buffer_merge (faac->cache, inbuf); } 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); } return; } /* create the frame */ if (in_size > size) { /* 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; subbuf = gst_buffer_merge (faac->cache, subbuf); 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); } outbuf = gst_buffer_new_and_alloc (faac->bytes); 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 (GST_ELEMENT (faac), "Error during AAC encoding"); gst_buffer_unref (inbuf); gst_buffer_unref (subbuf); return; } 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; } gst_pad_push (faac->srcpad, GST_DATA (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); } } static void gst_faac_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) { GstFaac *faac = GST_FAAC (object); 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; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_faac_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) { GstFaac *faac = GST_FAAC (object); 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; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstElementStateReturn gst_faac_change_state (GstElement *element) { GstFaac *faac = GST_FAAC (element); switch (GST_STATE_TRANSITION (element)) { case GST_STATE_PAUSED_TO_READY: 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; break; default: break; } if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element); return GST_STATE_SUCCESS; } static gboolean plugin_init (GstPlugin *plugin) { return gst_element_register (plugin, "faac", GST_RANK_NONE, GST_TYPE_FAAC); } GST_PLUGIN_DEFINE ( GST_VERSION_MAJOR, GST_VERSION_MINOR, "faac", "Free AAC Encoder (FAAC)", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN )