/* GStreamer FAAC (Free AAC Encoder) plugin * Copyright (C) 2003 Ronald Bultje * Copyright (C) 2009 Mark Nauwelaerts * * 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. */ /** * SECTION:element-faac * @see_also: faad * * faac encodes raw audio to AAC (MPEG-4 part 3) streams. * * * The #GstFaac:profile property determines the AAC profile, where the default * LC (Low Complexity) profile is most widely used, supported and suitable for * general use. The other profiles are very rarely used and often not supported. * * * Example launch line * |[ * gst-launch 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 #include #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 ], " \ "stream-format = (string) { adts, raw } " 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)); enum { ARG_0, 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_finalize (GObject * object); static void gst_faac_reset (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_configure_source_pad (GstFaac * faac); static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps); static GstCaps *gst_faac_sink_getcaps (GstPad * pad); static GstFlowReturn gst_faac_push_buffers (GstFaac * faac, gboolean force); 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 #define FAAC_DEFAULT_OUTPUTFORMAT 0 /* RAW */ #define FAAC_DEFAULT_BITRATE 128 * 1000 #define FAAC_DEFAULT_PROFILE LOW #define FAAC_DEFAULT_TNS FALSE #define FAAC_DEFAULT_MIDSIDE TRUE #define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL 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_simple (element_class, "AAC audio encoder", "Codec/Encoder/Audio", "Free MPEG-2/4 AAC encoder", "Ronald Bultje "); 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; } 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; gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_faac_finalize); /* 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, FAAC_DEFAULT_BITRATE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, ARG_PROFILE, g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile", GST_TYPE_FAAC_PROFILE, FAAC_DEFAULT_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, ARG_TNS, g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping", FAAC_DEFAULT_TNS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, ARG_MIDSIDE, g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding", FAAC_DEFAULT_MIDSIDE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, ARG_SHORTCTL, g_param_spec_enum ("shortctl", "Block type", "Block type encorcing", GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /* virtual functions */ gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state); } static void gst_faac_init (GstFaac * faac) { 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_getcaps_function (faac->sinkpad, GST_DEBUG_FUNCPTR (gst_faac_sink_getcaps)); 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); faac->adapter = gst_adapter_new (); /* default properties */ faac->bitrate = FAAC_DEFAULT_BITRATE; faac->profile = FAAC_DEFAULT_PROFILE; faac->shortctl = FAAC_DEFAULT_SHORTCTL; faac->outputformat = FAAC_DEFAULT_OUTPUTFORMAT; faac->tns = FAAC_DEFAULT_TNS; faac->midside = FAAC_DEFAULT_MIDSIDE; gst_faac_reset (faac); } static void gst_faac_reset (GstFaac * faac) { faac->handle = NULL; faac->samplerate = -1; faac->channels = -1; faac->offset = 0; gst_adapter_clear (faac->adapter); } static void gst_faac_finalize (GObject * object) { GstFaac *faac = (GstFaac *) object; g_object_unref (faac->adapter); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_faac_close_encoder (GstFaac * faac) { if (faac->handle) faacEncClose (faac->handle); faac->handle = NULL; gst_adapter_clear (faac->adapter); faac->offset = 0; } static const GstAudioChannelPosition aac_channel_positions[][8] = { {GST_AUDIO_CHANNEL_POSITION_FRONT_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_LFE} }; static GstCaps * gst_faac_sink_getcaps (GstPad * pad) { static volatile gsize sinkcaps = 0; if (g_once_init_enter (&sinkcaps)) { GstCaps *tmp = gst_caps_new_empty (); GstStructure *s, *t; gint i, c; s = gst_structure_new ("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", GST_TYPE_INT_RANGE, 8000, 96000, NULL); for (i = 1; i <= 6; i++) { GValue chanpos = { 0 }; GValue pos = { 0 }; t = gst_structure_copy (s); gst_structure_set (t, "channels", G_TYPE_INT, i, NULL); g_value_init (&chanpos, GST_TYPE_ARRAY); g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION); for (c = 0; c < i; c++) { g_value_set_enum (&pos, aac_channel_positions[i - 1][c]); gst_value_array_append_value (&chanpos, &pos); } g_value_unset (&pos); gst_structure_set_value (t, "channel-positions", &chanpos); g_value_unset (&chanpos); gst_caps_append_structure (tmp, t); } gst_structure_free (s); GST_DEBUG_OBJECT (pad, "Generated sinkcaps: %" GST_PTR_FORMAT, tmp); g_once_init_leave (&sinkcaps, (gsize) tmp); } return gst_caps_ref ((GstCaps *) sinkcaps); } /* check downstream caps to configure format */ static void gst_faac_negotiate (GstFaac * faac) { GstCaps *caps; caps = gst_pad_get_allowed_caps (faac->srcpad); 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; 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 (caps) gst_caps_unref (caps); } 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 refuse_caps; if (!gst_structure_get_int (structure, "channels", &channels) || !gst_structure_get_int (structure, "rate", &samplerate)) { goto refuse_caps; } 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) goto refuse_caps; /* If the encoder is initialized, do not reinitialize it again if not necessary */ if (faac->handle) { if (samplerate == faac->samplerate && channels == faac->channels && fmt == faac->format) return TRUE; /* clear out pending frames */ gst_faac_push_buffers (faac, TRUE); gst_faac_close_encoder (faac); } if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes))) goto setup_failed; /* ok, record and set up */ faac->format = fmt; faac->bps = bps; faac->handle = handle; faac->bytes = bytes; faac->samples = samples; faac->channels = channels; faac->samplerate = samplerate; gst_faac_negotiate (faac); /* finish up */ result = gst_faac_configure_source_pad (faac); done: gst_object_unref (faac); return result; /* ERRORS */ setup_failed: { GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL)); goto done; } refuse_caps: { GST_WARNING_OBJECT (faac, "refused caps %" GST_PTR_FORMAT, caps); goto done; } } static gboolean gst_faac_configure_source_pad (GstFaac * faac) { GstCaps *allowed_caps; GstCaps *srccaps; gboolean ret = FALSE; gint n, ver, mpegversion = 2; 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) { 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; } /* default 0 to start with, libfaac chooses based on bitrate */ conf->bandWidth = 0; if (!faacEncSetConfiguration (faac->handle, conf)) goto set_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 * faac->channels); GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual); GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth); /* 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, "stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"), NULL); if (!faac->outputformat) { 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); free (config); /* 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 GstFlowReturn gst_faac_push_buffers (GstFaac * faac, gboolean force) { GstFlowReturn ret = GST_FLOW_OK; gint av, frame_size, size, ret_size; GstBuffer *outbuf; guint64 timestamp, distance; const guint8 *data; /* samples already considers channel count */ frame_size = faac->samples * faac->bps; while (G_LIKELY (ret == GST_FLOW_OK)) { av = gst_adapter_available (faac->adapter); GST_LOG_OBJECT (faac, "pushing: force: %d, frame_size: %d, av: %d, " "offset: %d", force, frame_size, av, faac->offset); /* idea: * - start of adapter corresponds with what has already been encoded * (i.e. really returned by faac) * - start + offset is what needs to be fed to faac next * That way we can timestamp the output based * on adapter provided timestamp (and duration is a fixed frame duration) */ /* not enough data for one frame and no flush forcing */ if (!force && (av < frame_size + faac->offset)) break; if (G_LIKELY (av - faac->offset >= frame_size)) { GST_LOG_OBJECT (faac, "encoding a frame"); data = gst_adapter_peek (faac->adapter, faac->offset + frame_size); data += faac->offset; size = frame_size; } else if (av - faac->offset > 0) { GST_LOG_OBJECT (faac, "encoding leftover"); data = gst_adapter_peek (faac->adapter, av); data += faac->offset; size = av - faac->offset; } else { GST_LOG_OBJECT (faac, "emptying encoder"); data = NULL; size = 0; } outbuf = gst_buffer_new_and_alloc (faac->bytes); if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data, size / faac->bps, GST_BUFFER_DATA (outbuf), faac->bytes)) < 0)) { gst_buffer_unref (outbuf); goto encode_failed; } GST_LOG_OBJECT (faac, "encoder return: %d", ret_size); /* consumed, advanced view */ faac->offset += size; g_assert (faac->offset <= av); if (G_UNLIKELY (!ret_size)) { gst_buffer_unref (outbuf); if (size) continue; else break; } /* deal with encoder lead-out */ if (G_UNLIKELY (av == 0 && faac->offset == 0)) { GST_DEBUG_OBJECT (faac, "encoder returned additional data"); /* continuous with previous output, ok to have 0 duration */ timestamp = faac->next_ts; } else { /* after some caching, finally some data */ /* adapter gives time */ timestamp = gst_adapter_prev_timestamp (faac->adapter, &distance); } if (G_LIKELY ((av = gst_adapter_available (faac->adapter)) >= frame_size)) { /* must have then come from a complete frame */ gst_adapter_flush (faac->adapter, frame_size); faac->offset -= frame_size; size = frame_size; } else { /* otherwise leftover */ gst_adapter_clear (faac->adapter); faac->offset = 0; size = av; } GST_BUFFER_SIZE (outbuf) = ret_size; if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp))) GST_BUFFER_TIMESTAMP (outbuf) = timestamp + GST_FRAMES_TO_CLOCK_TIME (distance / faac->channels / faac->bps, faac->samplerate); GST_BUFFER_DURATION (outbuf) = GST_FRAMES_TO_CLOCK_TIME (size / faac->channels / faac->bps, faac->samplerate); faac->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); /* perhaps check/set DISCONT based on timestamps ? */ GST_LOG_OBJECT (faac, "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))); gst_buffer_set_caps (outbuf, GST_PAD_CAPS (faac->srcpad)); ret = gst_pad_push (faac->srcpad, outbuf); } /* in case encoder returns less than expected, clear our view as well */ if (G_UNLIKELY (force)) { #ifndef GST_DISABLE_GST_DEBUG if ((av = gst_adapter_available (faac->adapter))) GST_WARNING_OBJECT (faac, "encoder left %d bytes; discarding", av); #endif gst_adapter_clear (faac->adapter); faac->offset = 0; } return ret; /* ERRORS */ encode_failed: { GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL)); return GST_FLOW_ERROR; } } static gboolean gst_faac_sink_event (GstPad * pad, GstEvent * event) { GstFaac *faac; gboolean ret; faac = GST_FAAC (gst_pad_get_parent (pad)); GST_LOG_OBJECT (faac, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: { if (faac->handle) { /* flush first */ GST_DEBUG_OBJECT (faac, "Pushing out remaining buffers because of EOS"); gst_faac_push_buffers (faac, TRUE); } 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; GstFaac *faac; faac = GST_FAAC (gst_pad_get_parent (pad)); if (!faac->handle) goto no_handle; GST_LOG_OBJECT (faac, "Got buffer time: %" GST_TIME_FORMAT " duration: %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf))); gst_adapter_push (faac->adapter, inbuf); result = gst_faac_push_buffers (faac, FALSE); 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; } } 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; 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; 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_faac_close_encoder (faac); gst_faac_reset (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)