/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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-flacenc * @title: flacenc * @see_also: #GstFlacDec * * flacenc encodes FLAC streams. * [FLAC](http://flac.sourceforge.net/) is a Free Lossless Audio Codec. * FLAC audio can directly be written into a file, or embedded into containers * such as oggmux or matroskamux. * * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac * ]| Encode a short sine wave into FLAC * |[ * gst-launch-1.0 cdparanoiasrc mode=continuous ! queue ! audioconvert ! flacenc ! filesink location=cd.flac * ]| Rip a whole audio CD into a single FLAC file, with the track table saved as a CUE sheet inside the FLAC file * |[ * gst-launch-1.0 cdparanoiasrc track=5 ! queue ! audioconvert ! flacenc ! filesink location=track5.flac * ]| Rip track 5 of an audio CD and encode it losslessly to a FLAC file * */ /* TODO: - We currently don't handle discontinuities in the stream in a useful * way and instead rely on the developer plugging in audiorate if * the stream contains discontinuities. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include #include "gstflacelements.h" /* Taken from http://flac.sourceforge.net/format.html#frame_header */ static const GstAudioChannelPosition channel_positions[8][8] = { {GST_AUDIO_CHANNEL_POSITION_MONO}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { 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_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE1, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, /* FIXME: 7/8 channel layouts are not defined in the FLAC specs */ { 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_LFE1, GST_AUDIO_CHANNEL_POSITION_REAR_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_LFE1, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT} }; static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-flac") ); enum { PROP_0, PROP_QUALITY, PROP_STREAMABLE_SUBSET, PROP_MID_SIDE_STEREO, PROP_LOOSE_MID_SIDE_STEREO, PROP_BLOCKSIZE, PROP_MAX_LPC_ORDER, PROP_QLP_COEFF_PRECISION, PROP_QLP_COEFF_PREC_SEARCH, PROP_ESCAPE_CODING, PROP_EXHAUSTIVE_MODEL_SEARCH, PROP_MIN_RESIDUAL_PARTITION_ORDER, PROP_MAX_RESIDUAL_PARTITION_ORDER, PROP_RICE_PARAMETER_SEARCH_DIST, PROP_PADDING, PROP_SEEKPOINTS }; GST_DEBUG_CATEGORY_STATIC (flacenc_debug); #define GST_CAT_DEFAULT flacenc_debug #define gst_flac_enc_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstFlacEnc, gst_flac_enc, GST_TYPE_AUDIO_ENCODER, G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL) G_IMPLEMENT_INTERFACE (GST_TYPE_TOC_SETTER, NULL) ); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (flacenc, "flacenc", GST_RANK_PRIMARY, GST_TYPE_FLAC_ENC, flac_element_init (plugin)); static gboolean gst_flac_enc_start (GstAudioEncoder * enc); static gboolean gst_flac_enc_stop (GstAudioEncoder * enc); static gboolean gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info); static GstFlowReturn gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf); static GstCaps *gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter); static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event); static gboolean gst_flac_enc_sink_query (GstAudioEncoder * enc, GstQuery * query); static void gst_flac_enc_finalize (GObject * object); static GstCaps *gst_flac_enc_generate_sink_caps (void); static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc, gint quality); static void gst_flac_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_flac_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static FLAC__StreamEncoderWriteStatus gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder, const FLAC__byte buffer[], size_t bytes, unsigned samples, unsigned current_frame, void *client_data); static FLAC__StreamEncoderSeekStatus gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder, FLAC__uint64 absolute_byte_offset, void *client_data); static FLAC__StreamEncoderTellStatus gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder, FLAC__uint64 * absolute_byte_offset, void *client_data); typedef struct { gboolean exhaustive_model_search; gboolean escape_coding; gboolean mid_side; gboolean loose_mid_side; guint qlp_coeff_precision; gboolean qlp_coeff_prec_search; guint min_residual_partition_order; guint max_residual_partition_order; guint rice_parameter_search_dist; guint max_lpc_order; guint blocksize; } GstFlacEncParams; static const GstFlacEncParams flacenc_params[] = { {FALSE, FALSE, FALSE, FALSE, 0, FALSE, 2, 2, 0, 0, 1152}, {FALSE, FALSE, TRUE, TRUE, 0, FALSE, 2, 2, 0, 0, 1152}, {FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 3, 0, 0, 1152}, {FALSE, FALSE, FALSE, FALSE, 0, FALSE, 3, 3, 0, 6, 4608}, {FALSE, FALSE, TRUE, TRUE, 0, FALSE, 3, 3, 0, 8, 4608}, {FALSE, FALSE, TRUE, FALSE, 0, FALSE, 3, 3, 0, 8, 4608}, {FALSE, FALSE, TRUE, FALSE, 0, FALSE, 0, 4, 0, 8, 4608}, {TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 8, 4608}, {TRUE, FALSE, TRUE, FALSE, 0, FALSE, 0, 6, 0, 12, 4608}, {TRUE, TRUE, TRUE, FALSE, 0, FALSE, 0, 16, 0, 32, 4608}, }; #define DEFAULT_QUALITY 5 #define DEFAULT_PADDING 0 #define DEFAULT_SEEKPOINTS -10 #define GST_TYPE_FLAC_ENC_QUALITY (gst_flac_enc_quality_get_type ()) static GType gst_flac_enc_quality_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {0, "0 - Fastest compression", "0"}, {1, "1", "1"}, {2, "2", "2"}, {3, "3", "3"}, {4, "4", "4"}, {5, "5 - Default", "5"}, {6, "6", "6"}, {7, "7", "7"}, {8, "8 - Highest compression", "8"}, {9, "9 - Insane", "9"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstFlacEncQuality", values); } return qtype; } static void gst_flac_enc_class_init (GstFlacEncClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstAudioEncoderClass *base_class; GstCaps *sink_caps; GstPadTemplate *sink_templ; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; base_class = (GstAudioEncoderClass *) (klass); GST_DEBUG_CATEGORY_INIT (flacenc_debug, "flacenc", 0, "Flac encoding element"); gobject_class->set_property = gst_flac_enc_set_property; gobject_class->get_property = gst_flac_enc_get_property; gobject_class->finalize = gst_flac_enc_finalize; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY, g_param_spec_enum ("quality", "Quality", "Speed versus compression tradeoff", GST_TYPE_FLAC_ENC_QUALITY, DEFAULT_QUALITY, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STREAMABLE_SUBSET, g_param_spec_boolean ("streamable-subset", "Streamable subset", "true to limit encoder to generating a Subset stream, else false", TRUE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MID_SIDE_STEREO, g_param_spec_boolean ("mid-side-stereo", "Do mid side stereo", "Do mid side stereo (only for stereo input)", flacenc_params[DEFAULT_QUALITY].mid_side, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LOOSE_MID_SIDE_STEREO, g_param_spec_boolean ("loose-mid-side-stereo", "Loose mid side stereo", "Loose mid side stereo", flacenc_params[DEFAULT_QUALITY].loose_mid_side, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BLOCKSIZE, g_param_spec_uint ("blocksize", "Blocksize", "Blocksize in samples", FLAC__MIN_BLOCK_SIZE, FLAC__MAX_BLOCK_SIZE, flacenc_params[DEFAULT_QUALITY].blocksize, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_LPC_ORDER, g_param_spec_uint ("max-lpc-order", "Max LPC order", "Max LPC order; 0 => use only fixed predictors", 0, FLAC__MAX_LPC_ORDER, flacenc_params[DEFAULT_QUALITY].max_lpc_order, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QLP_COEFF_PRECISION, g_param_spec_uint ("qlp-coeff-precision", "QLP coefficients precision", "Precision in bits of quantized linear-predictor coefficients; 0 = automatic", 0, 32, flacenc_params[DEFAULT_QUALITY].qlp_coeff_precision, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QLP_COEFF_PREC_SEARCH, g_param_spec_boolean ("qlp-coeff-prec-search", "Do QLP coefficients precision search", "false = use qlp_coeff_precision, " "true = search around qlp_coeff_precision, take best", flacenc_params[DEFAULT_QUALITY].qlp_coeff_prec_search, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ESCAPE_CODING, g_param_spec_boolean ("escape-coding", "Do Escape coding", "search for escape codes in the entropy coding stage " "for slightly better compression", flacenc_params[DEFAULT_QUALITY].escape_coding, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_EXHAUSTIVE_MODEL_SEARCH, g_param_spec_boolean ("exhaustive-model-search", "Do exhaustive model search", "do exhaustive search of LP coefficient quantization (expensive!)", flacenc_params[DEFAULT_QUALITY].exhaustive_model_search, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MIN_RESIDUAL_PARTITION_ORDER, g_param_spec_uint ("min-residual-partition-order", "Min residual partition order", "Min residual partition order (above 4 doesn't usually help much)", 0, 16, flacenc_params[DEFAULT_QUALITY].min_residual_partition_order, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_RESIDUAL_PARTITION_ORDER, g_param_spec_uint ("max-residual-partition-order", "Max residual partition order", "Max residual partition order (above 4 doesn't usually help much)", 0, 16, flacenc_params[DEFAULT_QUALITY].max_residual_partition_order, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_RICE_PARAMETER_SEARCH_DIST, g_param_spec_uint ("rice-parameter-search-dist", "rice_parameter_search_dist", "0 = try only calc'd parameter k; else try all [k-dist..k+dist] " "parameters, use best", 0, FLAC__MAX_RICE_PARTITION_ORDER, flacenc_params[DEFAULT_QUALITY].rice_parameter_search_dist, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PADDING, g_param_spec_uint ("padding", "Padding", "Write a PADDING block with this length in bytes", 0, G_MAXUINT, DEFAULT_PADDING, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEEKPOINTS, g_param_spec_int ("seekpoints", "Seekpoints", "Add SEEKTABLE metadata (if > 0, number of entries, if < 0, interval in sec)", -G_MAXINT, G_MAXINT, DEFAULT_SEEKPOINTS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &src_factory); sink_caps = gst_flac_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, "FLAC audio encoder", "Codec/Encoder/Audio", "Encodes audio with the FLAC lossless audio encoder", "Wim Taymans "); base_class->start = GST_DEBUG_FUNCPTR (gst_flac_enc_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_flac_enc_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_enc_set_format); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_flac_enc_handle_frame); base_class->getcaps = GST_DEBUG_FUNCPTR (gst_flac_enc_getcaps); base_class->sink_event = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event); base_class->sink_query = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_query); gst_type_mark_as_plugin_api (GST_TYPE_FLAC_ENC_QUALITY, 0); } static void gst_flac_enc_init (GstFlacEnc * flacenc) { GstAudioEncoder *enc = GST_AUDIO_ENCODER (flacenc); flacenc->encoder = FLAC__stream_encoder_new (); gst_flac_enc_update_quality (flacenc, DEFAULT_QUALITY); /* arrange granulepos marking (and required perfect ts) */ gst_audio_encoder_set_mark_granule (enc, TRUE); gst_audio_encoder_set_perfect_timestamp (enc, TRUE); } static void gst_flac_enc_finalize (GObject * object) { GstFlacEnc *flacenc = GST_FLAC_ENC (object); FLAC__stream_encoder_delete (flacenc->encoder); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_flac_enc_start (GstAudioEncoder * enc) { GstFlacEnc *flacenc = GST_FLAC_ENC (enc); GST_DEBUG_OBJECT (enc, "start"); flacenc->stopped = TRUE; flacenc->got_headers = FALSE; flacenc->last_flow = GST_FLOW_OK; flacenc->offset = 0; flacenc->eos = FALSE; flacenc->tags = gst_tag_list_new_empty (); flacenc->toc = NULL; flacenc->samples_in = 0; flacenc->samples_out = 0; return TRUE; } static gboolean gst_flac_enc_stop (GstAudioEncoder * enc) { GstFlacEnc *flacenc = GST_FLAC_ENC (enc); GST_DEBUG_OBJECT (enc, "stop"); gst_tag_list_unref (flacenc->tags); flacenc->tags = NULL; if (flacenc->toc) gst_toc_unref (flacenc->toc); flacenc->toc = NULL; if (FLAC__stream_encoder_get_state (flacenc->encoder) != FLAC__STREAM_ENCODER_UNINITIALIZED) { flacenc->stopped = TRUE; FLAC__stream_encoder_finish (flacenc->encoder); } if (flacenc->meta) { FLAC__metadata_object_delete (flacenc->meta[0]); if (flacenc->meta[1]) FLAC__metadata_object_delete (flacenc->meta[1]); if (flacenc->meta[2]) FLAC__metadata_object_delete (flacenc->meta[2]); if (flacenc->meta[3]) FLAC__metadata_object_delete (flacenc->meta[3]); g_free (flacenc->meta); flacenc->meta = NULL; } g_list_foreach (flacenc->headers, (GFunc) gst_mini_object_unref, NULL); g_list_free (flacenc->headers); flacenc->headers = NULL; gst_tag_setter_reset_tags (GST_TAG_SETTER (enc)); gst_toc_setter_reset (GST_TOC_SETTER (enc)); return TRUE; } static void add_one_tag (const GstTagList * list, const gchar * tag, gpointer user_data) { GList *comments; GList *it; GstFlacEnc *flacenc = GST_FLAC_ENC (user_data); /* IMAGE and PREVIEW_IMAGE tags are already written * differently, no need to store them inside the * vorbiscomments too */ if (strcmp (tag, GST_TAG_IMAGE) == 0 || strcmp (tag, GST_TAG_PREVIEW_IMAGE) == 0) return; comments = gst_tag_to_vorbis_comments (list, tag); for (it = comments; it != NULL; it = it->next) { FLAC__StreamMetadata_VorbisComment_Entry commment_entry; commment_entry.length = strlen (it->data); commment_entry.entry = it->data; FLAC__metadata_object_vorbiscomment_insert_comment (flacenc->meta[0], flacenc->meta[0]->data.vorbis_comment.num_comments, commment_entry, TRUE); g_free (it->data); } g_list_free (comments); } static gboolean add_cuesheet (const GstToc * toc, guint sample_rate, FLAC__StreamMetadata * cuesheet) { guint8 track_num = 0; gint64 start, stop; gchar *isrc = NULL; const gchar *is_legal; GList *list; GstTagList *tags; GstTocEntry *entry, *subentry = NULL; FLAC__StreamMetadata_CueSheet *cs; FLAC__StreamMetadata_CueSheet_Track *track; cs = &cuesheet->data.cue_sheet; if (!cs) return FALSE; /* check if the TOC entries is valid */ list = gst_toc_get_entries (toc); entry = list->data; if (gst_toc_entry_is_alternative (entry)) { list = gst_toc_entry_get_sub_entries (entry); while (list) { subentry = list->data; if (!gst_toc_entry_is_sequence (subentry)) return FALSE; list = g_list_next (list); } list = gst_toc_entry_get_sub_entries (entry); } if (gst_toc_entry_is_sequence (entry)) { while (list) { entry = list->data; if (!gst_toc_entry_is_sequence (entry)) return FALSE; list = g_list_next (list); } list = gst_toc_get_entries (toc); } /* add tracks in cuesheet */ while (list) { entry = list->data; gst_toc_entry_get_start_stop_times (entry, &start, &stop); tags = gst_toc_entry_get_tags (entry); if (tags) gst_tag_list_get_string (tags, GST_TAG_ISRC, &isrc); track = FLAC__metadata_object_cuesheet_track_new (); track->offset = (FLAC__uint64) gst_util_uint64_scale_round (start, sample_rate, GST_SECOND); track->number = (FLAC__byte) track_num + 1; if (isrc != NULL && strlen (isrc) <= 12) g_strlcpy (track->isrc, isrc, 13); if (track->number <= 0) return FALSE; if (!FLAC__metadata_object_cuesheet_insert_track (cuesheet, track_num, track, FALSE)) return FALSE; if (!FLAC__metadata_object_cuesheet_track_insert_blank_index (cuesheet, track_num, 0)) return FALSE; track_num++; list = g_list_next (list); } if (cs->num_tracks <= 0) return FALSE; /* add lead-out track in cuesheet */ track = FLAC__metadata_object_cuesheet_track_new (); track->offset = (FLAC__uint64) gst_util_uint64_scale_round (stop, sample_rate, GST_SECOND); track->number = 255; if (!FLAC__metadata_object_cuesheet_insert_track (cuesheet, cs->num_tracks, track, FALSE)) return FALSE; /* check if the cuesheet is valid */ if (!FLAC__metadata_object_cuesheet_is_legal (cuesheet, FALSE, &is_legal)) { g_warning ("%s\n", is_legal); return FALSE; } return TRUE; } static void gst_flac_enc_set_metadata (GstFlacEnc * flacenc, GstAudioInfo * info, guint64 total_samples) { const GstTagList *user_tags; GstTagList *copy; gint entries = 1; gint n_images, n_preview_images; FLAC__StreamMetadata *cuesheet; g_return_if_fail (flacenc != NULL); user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (flacenc)); if ((flacenc->tags == NULL) && (user_tags == NULL)) { return; } copy = gst_tag_list_merge (user_tags, flacenc->tags, gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (flacenc))); n_images = gst_tag_list_get_tag_size (copy, GST_TAG_IMAGE); n_preview_images = gst_tag_list_get_tag_size (copy, GST_TAG_PREVIEW_IMAGE); flacenc->meta = g_new0 (FLAC__StreamMetadata *, 4 + n_images + n_preview_images); flacenc->meta[0] = FLAC__metadata_object_new (FLAC__METADATA_TYPE_VORBIS_COMMENT); gst_tag_list_foreach (copy, add_one_tag, flacenc); if (!flacenc->toc) flacenc->toc = gst_toc_setter_get_toc (GST_TOC_SETTER (flacenc)); if (flacenc->toc) { cuesheet = FLAC__metadata_object_new (FLAC__METADATA_TYPE_CUESHEET); if (add_cuesheet (flacenc->toc, GST_AUDIO_INFO_RATE (info), cuesheet)) { flacenc->meta[entries] = cuesheet; entries++; } else { FLAC__metadata_object_delete (cuesheet); flacenc->meta[entries] = NULL; } } if (n_images + n_preview_images > 0) { GstSample *sample; GstBuffer *buffer; GstCaps *caps; const GstStructure *structure; GstTagImageType image_type = GST_TAG_IMAGE_TYPE_NONE; gint i, width = 0, height = 0, png_icon_count = 0, other_icon_count = 0; GstMapInfo map; for (i = 0; i < n_images + n_preview_images; i++) { gboolean is_preview_image = (i >= n_images); if (i < n_images) { if (!gst_tag_list_get_sample_index (copy, GST_TAG_IMAGE, i, &sample)) continue; } else { if (!gst_tag_list_get_sample_index (copy, GST_TAG_PREVIEW_IMAGE, i - n_images, &sample)) continue; } structure = gst_sample_get_info (sample); caps = gst_sample_get_caps (sample); if (!caps) { GST_FIXME_OBJECT (flacenc, "Image tag without caps"); gst_sample_unref (sample); continue; } flacenc->meta[entries] = FLAC__metadata_object_new (FLAC__METADATA_TYPE_PICTURE); GST_LOG_OBJECT (flacenc, "image info: %" GST_PTR_FORMAT, structure); if (structure) gst_structure_get (structure, "image-type", GST_TYPE_TAG_IMAGE_TYPE, &image_type, NULL); else image_type = GST_TAG_IMAGE_TYPE_NONE; GST_LOG_OBJECT (flacenc, "image caps: %" GST_PTR_FORMAT, caps); structure = gst_caps_get_structure (caps, 0); gst_structure_get (structure, "width", G_TYPE_INT, &width, "height", G_TYPE_INT, &height, NULL); /* Convert to ID3v2 APIC image type */ if (image_type == GST_TAG_IMAGE_TYPE_NONE) { if (is_preview_image) { /* 1 - 32x32 pixels 'file icon' (PNG only) * 2 - Other file icon * There may only be one each of picture type 1 and 2 in a file. */ if (width == 32 && height == 32 && gst_structure_has_name (structure, "image/png") && png_icon_count++ == 0) { image_type = 1; } else if (width <= 32 && height <= 32 && other_icon_count++ == 0) { image_type = 2; } else { image_type = 0; /* Other */ } } else { image_type = 0; /* Other */ } } else { /* GStreamer enum is the same but without the two icon types 1+2 */ image_type = image_type + 2; } buffer = gst_sample_get_buffer (sample); gst_buffer_map (buffer, &map, GST_MAP_READ); FLAC__metadata_object_picture_set_data (flacenc->meta[entries], map.data, map.size, TRUE); gst_buffer_unmap (buffer, &map); GST_LOG_OBJECT (flacenc, "Setting picture type %d", image_type); switch (image_type) { case GST_TAG_IMAGE_TYPE_NONE: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_OTHER; break; case GST_TAG_IMAGE_TYPE_FRONT_COVER: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_FRONT_COVER; break; case GST_TAG_IMAGE_TYPE_BACK_COVER: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_BACK_COVER; break; case GST_TAG_IMAGE_TYPE_LEAFLET_PAGE: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_LEAFLET_PAGE; break; case GST_TAG_IMAGE_TYPE_MEDIUM: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_MEDIA; break; case GST_TAG_IMAGE_TYPE_LEAD_ARTIST: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_LEAD_ARTIST; break; case GST_TAG_IMAGE_TYPE_ARTIST: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_ARTIST; break; case GST_TAG_IMAGE_TYPE_CONDUCTOR: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_CONDUCTOR; break; case GST_TAG_IMAGE_TYPE_BAND_ORCHESTRA: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_BAND; break; case GST_TAG_IMAGE_TYPE_COMPOSER: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_COMPOSER; break; case GST_TAG_IMAGE_TYPE_LYRICIST: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_LYRICIST; break; case GST_TAG_IMAGE_TYPE_RECORDING_LOCATION: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_RECORDING_LOCATION; break; case GST_TAG_IMAGE_TYPE_DURING_RECORDING: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_DURING_RECORDING; break; case GST_TAG_IMAGE_TYPE_DURING_PERFORMANCE: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_DURING_PERFORMANCE; break; case GST_TAG_IMAGE_TYPE_VIDEO_CAPTURE: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_VIDEO_SCREEN_CAPTURE; break; case GST_TAG_IMAGE_TYPE_FISH: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_FISH; break; case GST_TAG_IMAGE_TYPE_ILLUSTRATION: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_ILLUSTRATION; break; case GST_TAG_IMAGE_TYPE_BAND_ARTIST_LOGO: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_BAND_LOGOTYPE; break; case GST_TAG_IMAGE_TYPE_PUBLISHER_STUDIO_LOGO: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_PUBLISHER_LOGOTYPE; break; case GST_TAG_IMAGE_TYPE_UNDEFINED: default: flacenc->meta[entries]->data.picture.type = FLAC__STREAM_METADATA_PICTURE_TYPE_UNDEFINED; break; } if (width > 0 && height > 0) { flacenc->meta[entries]->data.picture.width = width; flacenc->meta[entries]->data.picture.height = height; } FLAC__metadata_object_picture_set_mime_type (flacenc->meta[entries], (char *) gst_structure_get_name (structure), TRUE); gst_sample_unref (sample); entries++; } } if (flacenc->seekpoints && total_samples != GST_CLOCK_TIME_NONE) { gboolean res; guint samples; flacenc->meta[entries] = FLAC__metadata_object_new (FLAC__METADATA_TYPE_SEEKTABLE); if (flacenc->seekpoints > 0) { res = FLAC__metadata_object_seektable_template_append_spaced_points (flacenc->meta[entries], flacenc->seekpoints, total_samples); } else { samples = -flacenc->seekpoints * GST_AUDIO_INFO_RATE (info); res = FLAC__metadata_object_seektable_template_append_spaced_points_by_samples (flacenc->meta[entries], samples, total_samples); } if (!res) { GST_DEBUG_OBJECT (flacenc, "adding seekpoint template %d failed", flacenc->seekpoints); FLAC__metadata_object_delete (flacenc->meta[1]); flacenc->meta[entries] = NULL; } else { entries++; } } else if (flacenc->seekpoints && total_samples == GST_CLOCK_TIME_NONE) { GST_WARNING_OBJECT (flacenc, "total time unknown; can not add seekpoints"); } if (flacenc->padding > 0) { flacenc->meta[entries] = FLAC__metadata_object_new (FLAC__METADATA_TYPE_PADDING); flacenc->meta[entries]->length = flacenc->padding; entries++; } if (FLAC__stream_encoder_set_metadata (flacenc->encoder, flacenc->meta, entries) != true) g_warning ("Dude, i'm already initialized!"); gst_tag_list_unref (copy); } static GstCaps * gst_flac_enc_generate_sink_caps (void) { GstCaps *ret; gint i; GValue v_list = { 0, }; GValue v = { 0, }; GstStructure *s, *s2; g_value_init (&v_list, GST_TYPE_LIST); g_value_init (&v, G_TYPE_STRING); /* Use system's endianness */ g_value_set_static_string (&v, "S8"); gst_value_list_append_value (&v_list, &v); g_value_set_static_string (&v, GST_AUDIO_NE (S16)); gst_value_list_append_value (&v_list, &v); g_value_set_static_string (&v, GST_AUDIO_NE (S24)); gst_value_list_append_value (&v_list, &v); g_value_set_static_string (&v, GST_AUDIO_NE (S24_32)); gst_value_list_append_value (&v_list, &v); g_value_unset (&v); s = gst_structure_new_empty ("audio/x-raw"); gst_structure_take_value (s, "format", &v_list); gst_structure_set (s, "layout", G_TYPE_STRING, "interleaved", "rate", GST_TYPE_INT_RANGE, 1, 655350, NULL); ret = gst_caps_new_empty (); s2 = gst_structure_copy (s); gst_structure_set (s2, "channels", G_TYPE_INT, 1, NULL); gst_caps_append_structure (ret, s2); for (i = 2; i <= 8; i++) { guint64 channel_mask; s2 = gst_structure_copy (s); gst_audio_channel_positions_to_mask (channel_positions[i - 1], i, FALSE, &channel_mask); gst_structure_set (s2, "channels", G_TYPE_INT, i, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL); gst_caps_append_structure (ret, s2); } gst_structure_free (s); return ret; } static GstCaps * gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter) { GstCaps *ret = NULL, *caps = NULL; GstPad *pad; pad = GST_AUDIO_ENCODER_SINK_PAD (enc); ret = gst_pad_get_current_caps (pad); if (ret == NULL) { ret = gst_pad_get_pad_template_caps (pad); } GST_DEBUG_OBJECT (pad, "Return caps %" GST_PTR_FORMAT, ret); caps = gst_audio_encoder_proxy_getcaps (enc, ret, filter); gst_caps_unref (ret); return caps; } static guint64 gst_flac_enc_peer_query_total_samples (GstFlacEnc * flacenc, GstPad * pad, GstAudioInfo * info) { gint64 duration; GST_DEBUG_OBJECT (flacenc, "querying peer for DEFAULT format duration"); if (gst_pad_peer_query_duration (pad, GST_FORMAT_DEFAULT, &duration) && duration != GST_CLOCK_TIME_NONE) goto done; GST_DEBUG_OBJECT (flacenc, "querying peer for TIME format duration"); if (gst_pad_peer_query_duration (pad, GST_FORMAT_TIME, &duration) && duration != GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (flacenc, "peer reported duration %" GST_TIME_FORMAT, GST_TIME_ARGS (duration)); duration = GST_CLOCK_TIME_TO_FRAMES (duration, GST_AUDIO_INFO_RATE (info)); goto done; } GST_DEBUG_OBJECT (flacenc, "Upstream reported no total samples"); return GST_CLOCK_TIME_NONE; done: GST_DEBUG_OBJECT (flacenc, "Upstream reported %" G_GUINT64_FORMAT " total samples", duration); return duration; } static gint64 gst_flac_enc_get_latency (GstFlacEnc * flacenc) { /* The FLAC specification states that the data is processed in blocks, * regardless of the number of channels. Thus, The latency can be calculated * using the blocksize and rate. For example a 1 second block sampled at * 44.1KHz has a blocksize of 44100 */ /* Get the blocksize */ const guint blocksize = FLAC__stream_encoder_get_blocksize (flacenc->encoder); /* Get the sample rate in KHz */ const guint rate = FLAC__stream_encoder_get_sample_rate (flacenc->encoder); if (!rate) return 0; /* Calculate the latecy */ return (blocksize * GST_SECOND) / rate; } static gboolean gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info) { GstFlacEnc *flacenc; guint64 total_samples = GST_CLOCK_TIME_NONE; FLAC__StreamEncoderInitStatus init_status; flacenc = GST_FLAC_ENC (enc); /* if configured again, means something changed, can't handle that */ if (FLAC__stream_encoder_get_state (flacenc->encoder) != FLAC__STREAM_ENCODER_UNINITIALIZED) goto encoder_already_initialized; /* delay setting output caps/format until we have all headers */ gst_audio_get_channel_reorder_map (GST_AUDIO_INFO_CHANNELS (info), channel_positions[GST_AUDIO_INFO_CHANNELS (info) - 1], info->position, flacenc->channel_reorder_map); total_samples = gst_flac_enc_peer_query_total_samples (flacenc, GST_AUDIO_ENCODER_SINK_PAD (enc), info); FLAC__stream_encoder_set_bits_per_sample (flacenc->encoder, GST_AUDIO_INFO_DEPTH (info)); FLAC__stream_encoder_set_sample_rate (flacenc->encoder, GST_AUDIO_INFO_RATE (info)); FLAC__stream_encoder_set_channels (flacenc->encoder, GST_AUDIO_INFO_CHANNELS (info)); if (total_samples != GST_CLOCK_TIME_NONE) FLAC__stream_encoder_set_total_samples_estimate (flacenc->encoder, MIN (total_samples, G_GUINT64_CONSTANT (0x0FFFFFFFFF))); gst_flac_enc_set_metadata (flacenc, info, total_samples); /* callbacks clear to go now; * write callbacks receives headers during init */ flacenc->stopped = FALSE; init_status = FLAC__stream_encoder_init_stream (flacenc->encoder, gst_flac_enc_write_callback, gst_flac_enc_seek_callback, gst_flac_enc_tell_callback, NULL, flacenc); if (init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) goto failed_to_initialize; /* feedback to base class */ gst_audio_encoder_set_latency (enc, gst_flac_enc_get_latency (flacenc), gst_flac_enc_get_latency (flacenc)); return TRUE; encoder_already_initialized: { g_warning ("flac already initialized -- fixme allow this"); return FALSE; } failed_to_initialize: { GST_ELEMENT_ERROR (flacenc, LIBRARY, INIT, (NULL), ("could not initialize encoder (wrong parameters?) %d", init_status)); return FALSE; } } static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc, gint quality) { GstAudioInfo *info = gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (flacenc)); flacenc->quality = quality; #define DO_UPDATE(name, val, str) \ G_STMT_START { \ if (FLAC__stream_encoder_get_##name (flacenc->encoder) != \ flacenc_params[quality].val) { \ FLAC__stream_encoder_set_##name (flacenc->encoder, \ flacenc_params[quality].val); \ g_object_notify (G_OBJECT (flacenc), str); \ } \ } G_STMT_END g_object_freeze_notify (G_OBJECT (flacenc)); if (GST_AUDIO_INFO_CHANNELS (info) == 2 || GST_AUDIO_INFO_CHANNELS (info) == 0) { DO_UPDATE (do_mid_side_stereo, mid_side, "mid_side_stereo"); DO_UPDATE (loose_mid_side_stereo, loose_mid_side, "loose_mid_side"); } DO_UPDATE (blocksize, blocksize, "blocksize"); DO_UPDATE (max_lpc_order, max_lpc_order, "max_lpc_order"); DO_UPDATE (qlp_coeff_precision, qlp_coeff_precision, "qlp_coeff_precision"); DO_UPDATE (do_qlp_coeff_prec_search, qlp_coeff_prec_search, "qlp_coeff_prec_search"); DO_UPDATE (do_escape_coding, escape_coding, "escape_coding"); DO_UPDATE (do_exhaustive_model_search, exhaustive_model_search, "exhaustive_model_search"); DO_UPDATE (min_residual_partition_order, min_residual_partition_order, "min_residual_partition_order"); DO_UPDATE (max_residual_partition_order, max_residual_partition_order, "max_residual_partition_order"); DO_UPDATE (rice_parameter_search_dist, rice_parameter_search_dist, "rice_parameter_search_dist"); #undef DO_UPDATE g_object_thaw_notify (G_OBJECT (flacenc)); return TRUE; } static FLAC__StreamEncoderSeekStatus gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder, FLAC__uint64 absolute_byte_offset, void *client_data) { GstFlacEnc *flacenc; GstPad *peerpad; GstSegment seg; flacenc = GST_FLAC_ENC (client_data); if (flacenc->stopped) return FLAC__STREAM_ENCODER_SEEK_STATUS_OK; if ((peerpad = gst_pad_get_peer (GST_AUDIO_ENCODER_SRC_PAD (flacenc)))) { GstEvent *event; gboolean ret; GstQuery *query; gboolean seekable = FALSE; /* try to seek to the beginning of the output */ query = gst_query_new_seeking (GST_FORMAT_BYTES); if (gst_pad_query (peerpad, query)) { GstFormat format; gst_query_parse_seeking (query, &format, &seekable, NULL, NULL); if (format != GST_FORMAT_BYTES) seekable = FALSE; } else { GST_LOG_OBJECT (flacenc, "SEEKING query not handled"); } gst_query_unref (query); if (!seekable) { GST_DEBUG_OBJECT (flacenc, "downstream not seekable; not rewriting"); gst_object_unref (peerpad); return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED; } gst_segment_init (&seg, GST_FORMAT_BYTES); seg.start = absolute_byte_offset; seg.stop = GST_BUFFER_OFFSET_NONE; seg.time = 0; event = gst_event_new_segment (&seg); ret = gst_pad_send_event (peerpad, event); gst_object_unref (peerpad); if (ret) { GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " %s", (guint64) absolute_byte_offset, "succeeded"); } else { GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " %s", (guint64) absolute_byte_offset, "failed"); return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED; } } else { GST_DEBUG ("Seek to %" G_GUINT64_FORMAT " failed (no peer pad)", (guint64) absolute_byte_offset); } flacenc->offset = absolute_byte_offset; return FLAC__STREAM_ENCODER_SEEK_STATUS_OK; } static void notgst_value_array_append_buffer (GValue * array_val, GstBuffer * buf) { GValue value = { 0, }; g_value_init (&value, GST_TYPE_BUFFER); /* copy buffer to avoid problems with circular refcounts */ buf = gst_buffer_copy (buf); /* again, for good measure */ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER); gst_value_set_buffer (&value, buf); gst_buffer_unref (buf); gst_value_array_append_value (array_val, &value); g_value_unset (&value); } #define HDR_TYPE_STREAMINFO 0 #define HDR_TYPE_VORBISCOMMENT 4 static GstFlowReturn gst_flac_enc_process_stream_headers (GstFlacEnc * enc) { GstBuffer *vorbiscomment = NULL; GstBuffer *streaminfo = NULL; GstBuffer *marker = NULL; GValue array = { 0, }; GstCaps *caps; GList *l; GstFlowReturn ret = GST_FLOW_OK; GstAudioInfo *info = gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (enc)); caps = gst_caps_new_simple ("audio/x-flac", "channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info), "rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL); for (l = enc->headers; l != NULL; l = l->next) { GstBuffer *buf; GstMapInfo map; guint8 *data; gsize size; /* mark buffers so oggmux will ignore them if it already muxed the * header buffers from the streamheaders field in the caps */ l->data = gst_buffer_make_writable (GST_BUFFER_CAST (l->data)); buf = GST_BUFFER_CAST (l->data); GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER); gst_buffer_map (buf, &map, GST_MAP_READ); data = map.data; size = map.size; /* find initial 4-byte marker which we need to skip later on */ if (size == 4 && memcmp (data, "fLaC", 4) == 0) { marker = buf; } else if (size > 1 && (data[0] & 0x7f) == HDR_TYPE_STREAMINFO) { streaminfo = buf; } else if (size > 1 && (data[0] & 0x7f) == HDR_TYPE_VORBISCOMMENT) { vorbiscomment = buf; } gst_buffer_unmap (buf, &map); } if (marker == NULL || streaminfo == NULL || vorbiscomment == NULL) { GST_WARNING_OBJECT (enc, "missing header %p %p %p, muxing into container " "formats may be broken", marker, streaminfo, vorbiscomment); goto push_headers; } g_value_init (&array, GST_TYPE_ARRAY); /* add marker including STREAMINFO header */ { GstBuffer *buf; guint16 num; GstMapInfo map; guint8 *bdata; gsize slen; /* minus one for the marker that is merged with streaminfo here */ num = g_list_length (enc->headers) - 1; slen = gst_buffer_get_size (streaminfo); buf = gst_buffer_new_and_alloc (13 + slen); gst_buffer_map (buf, &map, GST_MAP_WRITE); bdata = map.data; bdata[0] = 0x7f; memcpy (bdata + 1, "FLAC", 4); bdata[5] = 0x01; /* mapping version major */ bdata[6] = 0x00; /* mapping version minor */ bdata[7] = (num & 0xFF00) >> 8; bdata[8] = (num & 0x00FF) >> 0; memcpy (bdata + 9, "fLaC", 4); gst_buffer_extract (streaminfo, 0, bdata + 13, slen); gst_buffer_unmap (buf, &map); notgst_value_array_append_buffer (&array, buf); gst_buffer_unref (buf); } /* add VORBISCOMMENT header */ notgst_value_array_append_buffer (&array, vorbiscomment); /* add other headers, if there are any */ for (l = enc->headers; l != NULL; l = l->next) { GstBuffer *buf = GST_BUFFER_CAST (l->data); if (buf != marker && buf != streaminfo && buf != vorbiscomment) { notgst_value_array_append_buffer (&array, buf); } } gst_structure_set_value (gst_caps_get_structure (caps, 0), "streamheader", &array); g_value_unset (&array); push_headers: gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps); gst_audio_encoder_set_headers (GST_AUDIO_ENCODER (enc), enc->headers); enc->headers = NULL; gst_caps_unref (caps); return ret; } static FLAC__StreamEncoderWriteStatus gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder, const FLAC__byte buffer[], size_t bytes, unsigned samples, unsigned current_frame, void *client_data) { GstFlowReturn ret = GST_FLOW_OK; GstFlacEnc *flacenc; GstBuffer *outbuf; GstSegment *segment; GstClockTime duration; flacenc = GST_FLAC_ENC (client_data); if (flacenc->stopped) return FLAC__STREAM_ENCODER_WRITE_STATUS_OK; outbuf = gst_buffer_new_and_alloc (bytes); gst_buffer_fill (outbuf, 0, buffer, bytes); /* we assume libflac passes us stuff neatly framed */ if (!flacenc->got_headers) { if (samples == 0) { GST_DEBUG_OBJECT (flacenc, "Got header, queueing (%u bytes)", (guint) bytes); flacenc->headers = g_list_append (flacenc->headers, outbuf); /* note: it's important that we increase our byte offset */ goto out; } else { GST_INFO_OBJECT (flacenc, "Non-header packet, we have all headers now"); ret = gst_flac_enc_process_stream_headers (flacenc); flacenc->got_headers = TRUE; } } if (flacenc->got_headers && samples == 0) { /* header fixup, push downstream directly */ GST_DEBUG_OBJECT (flacenc, "Fixing up headers at pos=%" G_GUINT64_FORMAT ", size=%u", flacenc->offset, (guint) bytes); #if 0 GST_MEMDUMP_OBJECT (flacenc, "Presumed header fragment", GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf)); #endif ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf); } else { /* regular frame data, pass to base class */ if (flacenc->eos && flacenc->samples_in == flacenc->samples_out + samples) { /* If encoding part of a frame, and we have no set stop time on * the output segment, we update the segment stop time to reflect * the last sample. This will let oggmux set the last page's * granpos to tell a decoder the dummy samples should be clipped. */ segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (flacenc); if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) { GST_DEBUG_OBJECT (flacenc, "No stop time and partial frame, updating segment"); duration = gst_util_uint64_scale (flacenc->samples_out + samples, GST_SECOND, FLAC__stream_encoder_get_sample_rate (flacenc->encoder)); segment->stop = segment->start + duration; GST_DEBUG_OBJECT (flacenc, "new output segment %" GST_SEGMENT_FORMAT, segment); gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (flacenc), gst_event_new_segment (segment)); } } GST_LOG ("Pushing buffer: samples=%u, size=%u, pos=%" G_GUINT64_FORMAT, samples, (guint) bytes, flacenc->offset); ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc), outbuf, samples); } if (ret != GST_FLOW_OK) GST_DEBUG_OBJECT (flacenc, "flow: %s", gst_flow_get_name (ret)); flacenc->last_flow = ret; out: flacenc->offset += bytes; if (ret != GST_FLOW_OK) return FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR; return FLAC__STREAM_ENCODER_WRITE_STATUS_OK; } static FLAC__StreamEncoderTellStatus gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder, FLAC__uint64 * absolute_byte_offset, void *client_data) { GstFlacEnc *flacenc = GST_FLAC_ENC (client_data); *absolute_byte_offset = flacenc->offset; return FLAC__STREAM_ENCODER_TELL_STATUS_OK; } static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event) { GstFlacEnc *flacenc; GstTagList *taglist; GstToc *toc; gboolean ret = FALSE; flacenc = GST_FLAC_ENC (enc); GST_DEBUG ("Received %s event on sinkpad, %" GST_PTR_FORMAT, GST_EVENT_TYPE_NAME (event), event); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: flacenc->eos = TRUE; ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); break; case GST_EVENT_TAG: if (flacenc->tags) { gst_event_parse_tag (event, &taglist); gst_tag_list_insert (flacenc->tags, taglist, gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (flacenc))); } else { g_assert_not_reached (); } ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); break; case GST_EVENT_TOC: gst_event_parse_toc (event, &toc, NULL); if (toc) { if (flacenc->toc != toc) { if (flacenc->toc) gst_toc_unref (flacenc->toc); flacenc->toc = toc; } } ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); break; case GST_EVENT_SEGMENT: flacenc->samples_in = 0; flacenc->samples_out = 0; ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); break; default: ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); break; } return ret; } static gboolean gst_flac_enc_sink_query (GstAudioEncoder * enc, GstQuery * query) { GstPad *pad = GST_AUDIO_ENCODER_SINK_PAD (enc); gboolean ret = FALSE; GST_DEBUG ("Received %s query on sinkpad, %" GST_PTR_FORMAT, GST_QUERY_TYPE_NAME (query), query); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_ACCEPT_CAPS:{ GstCaps *acceptable, *caps; acceptable = gst_pad_get_current_caps (pad); if (acceptable == NULL) { acceptable = gst_pad_get_pad_template_caps (pad); } gst_query_parse_accept_caps (query, &caps); gst_query_set_accept_caps_result (query, gst_caps_is_subset (caps, acceptable)); gst_caps_unref (acceptable); ret = TRUE; } break; default: ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_query (enc, query); break; } return ret; } #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define READ_INT24 GST_READ_UINT24_LE #else #define READ_INT24 GST_READ_UINT24_BE #endif static GstFlowReturn gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer) { GstFlacEnc *flacenc; FLAC__int32 *data; gint samples, width, channels; gulong i; gint j; FLAC__bool res; GstMapInfo map; GstAudioInfo *info = gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (enc)); gint *reorder_map; flacenc = GST_FLAC_ENC (enc); /* base class ensures configuration */ g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (info) != 0, GST_FLOW_NOT_NEGOTIATED); width = GST_AUDIO_INFO_WIDTH (info); channels = GST_AUDIO_INFO_CHANNELS (info); reorder_map = flacenc->channel_reorder_map; if (G_UNLIKELY (!buffer)) { if (flacenc->eos) { GST_DEBUG_OBJECT (flacenc, "finish encoding"); FLAC__stream_encoder_finish (flacenc->encoder); } else { /* can't handle intermittent draining/resyncing */ GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL), ("Stream discontinuity detected. " "The output may have wrong timestamps, " "consider using audiorate to handle discontinuities")); } return flacenc->last_flow; } gst_buffer_map (buffer, &map, GST_MAP_READ); samples = map.size / (width >> 3); data = g_malloc (samples * sizeof (FLAC__int32)); samples /= channels; GST_LOG_OBJECT (flacenc, "processing %d samples, %d channels", samples, channels); if (width == 8) { gint8 *indata = (gint8 *) map.data; for (i = 0; i < samples; i++) for (j = 0; j < channels; j++) data[i * channels + reorder_map[j]] = (FLAC__int32) indata[i * channels + j]; } else if (width == 16) { gint16 *indata = (gint16 *) map.data; for (i = 0; i < samples; i++) for (j = 0; j < channels; j++) data[i * channels + reorder_map[j]] = (FLAC__int32) indata[i * channels + j]; } else if (width == 24) { guint8 *indata = (guint8 *) map.data; guint32 val; for (i = 0; i < samples; i++) for (j = 0; j < channels; j++) { val = READ_INT24 (&indata[3 * (i * channels + j)]); if (val & 0x00800000) val |= 0xff000000; data[i * channels + reorder_map[j]] = (FLAC__int32) val; } } else if (width == 32) { gint32 *indata = (gint32 *) map.data; for (i = 0; i < samples; i++) for (j = 0; j < channels; j++) data[i * channels + reorder_map[j]] = (FLAC__int32) indata[i * channels + j]; } else { g_assert_not_reached (); } gst_buffer_unmap (buffer, &map); res = FLAC__stream_encoder_process_interleaved (flacenc->encoder, (const FLAC__int32 *) data, samples); flacenc->samples_in += samples; g_free (data); if (!res) { if (flacenc->last_flow == GST_FLOW_OK) return GST_FLOW_ERROR; else return flacenc->last_flow; } return GST_FLOW_OK; } static void gst_flac_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstFlacEnc *this = GST_FLAC_ENC (object); GstAudioEncoder *enc = GST_AUDIO_ENCODER (object); guint64 curr_latency = 0, old_latency = gst_flac_enc_get_latency (this); GST_OBJECT_LOCK (this); switch (prop_id) { case PROP_QUALITY: gst_flac_enc_update_quality (this, g_value_get_enum (value)); break; case PROP_STREAMABLE_SUBSET: FLAC__stream_encoder_set_streamable_subset (this->encoder, g_value_get_boolean (value)); break; case PROP_MID_SIDE_STEREO: FLAC__stream_encoder_set_do_mid_side_stereo (this->encoder, g_value_get_boolean (value)); break; case PROP_LOOSE_MID_SIDE_STEREO: FLAC__stream_encoder_set_loose_mid_side_stereo (this->encoder, g_value_get_boolean (value)); break; case PROP_BLOCKSIZE: FLAC__stream_encoder_set_blocksize (this->encoder, g_value_get_uint (value)); break; case PROP_MAX_LPC_ORDER: FLAC__stream_encoder_set_max_lpc_order (this->encoder, g_value_get_uint (value)); break; case PROP_QLP_COEFF_PRECISION: FLAC__stream_encoder_set_qlp_coeff_precision (this->encoder, g_value_get_uint (value)); break; case PROP_QLP_COEFF_PREC_SEARCH: FLAC__stream_encoder_set_do_qlp_coeff_prec_search (this->encoder, g_value_get_boolean (value)); break; case PROP_ESCAPE_CODING: FLAC__stream_encoder_set_do_escape_coding (this->encoder, g_value_get_boolean (value)); break; case PROP_EXHAUSTIVE_MODEL_SEARCH: FLAC__stream_encoder_set_do_exhaustive_model_search (this->encoder, g_value_get_boolean (value)); break; case PROP_MIN_RESIDUAL_PARTITION_ORDER: FLAC__stream_encoder_set_min_residual_partition_order (this->encoder, g_value_get_uint (value)); break; case PROP_MAX_RESIDUAL_PARTITION_ORDER: FLAC__stream_encoder_set_max_residual_partition_order (this->encoder, g_value_get_uint (value)); break; case PROP_RICE_PARAMETER_SEARCH_DIST: FLAC__stream_encoder_set_rice_parameter_search_dist (this->encoder, g_value_get_uint (value)); break; case PROP_PADDING: this->padding = g_value_get_uint (value); break; case PROP_SEEKPOINTS: this->seekpoints = g_value_get_int (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (this); /* Update latency if it has changed */ curr_latency = gst_flac_enc_get_latency (this); if (old_latency != curr_latency) gst_audio_encoder_set_latency (enc, curr_latency, curr_latency); } static void gst_flac_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstFlacEnc *this = GST_FLAC_ENC (object); GST_OBJECT_LOCK (this); switch (prop_id) { case PROP_QUALITY: g_value_set_enum (value, this->quality); break; case PROP_STREAMABLE_SUBSET: g_value_set_boolean (value, FLAC__stream_encoder_get_streamable_subset (this->encoder)); break; case PROP_MID_SIDE_STEREO: g_value_set_boolean (value, FLAC__stream_encoder_get_do_mid_side_stereo (this->encoder)); break; case PROP_LOOSE_MID_SIDE_STEREO: g_value_set_boolean (value, FLAC__stream_encoder_get_loose_mid_side_stereo (this->encoder)); break; case PROP_BLOCKSIZE: g_value_set_uint (value, FLAC__stream_encoder_get_blocksize (this->encoder)); break; case PROP_MAX_LPC_ORDER: g_value_set_uint (value, FLAC__stream_encoder_get_max_lpc_order (this->encoder)); break; case PROP_QLP_COEFF_PRECISION: g_value_set_uint (value, FLAC__stream_encoder_get_qlp_coeff_precision (this->encoder)); break; case PROP_QLP_COEFF_PREC_SEARCH: g_value_set_boolean (value, FLAC__stream_encoder_get_do_qlp_coeff_prec_search (this->encoder)); break; case PROP_ESCAPE_CODING: g_value_set_boolean (value, FLAC__stream_encoder_get_do_escape_coding (this->encoder)); break; case PROP_EXHAUSTIVE_MODEL_SEARCH: g_value_set_boolean (value, FLAC__stream_encoder_get_do_exhaustive_model_search (this->encoder)); break; case PROP_MIN_RESIDUAL_PARTITION_ORDER: g_value_set_uint (value, FLAC__stream_encoder_get_min_residual_partition_order (this->encoder)); break; case PROP_MAX_RESIDUAL_PARTITION_ORDER: g_value_set_uint (value, FLAC__stream_encoder_get_max_residual_partition_order (this->encoder)); break; case PROP_RICE_PARAMETER_SEARCH_DIST: g_value_set_uint (value, FLAC__stream_encoder_get_rice_parameter_search_dist (this->encoder)); break; case PROP_PADDING: g_value_set_uint (value, this->padding); break; case PROP_SEEKPOINTS: g_value_set_int (value, this->seekpoints); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (this); }