X-Git-Url: http://git.maemo.org/git/?p=mafwsubrenderer;a=blobdiff_plain;f=gst-plugins-base-subtitles0.10%2Fext%2Fvorbis%2Fgstvorbisenc.c;fp=gst-plugins-base-subtitles0.10%2Fext%2Fvorbis%2Fgstvorbisenc.c;h=9cdf484e1553a76d16e78c10c073b56d3b6aed01;hp=0000000000000000000000000000000000000000;hb=57ba96e291a055f69dbfd4ae9f1ae2390e36986e;hpb=be2c98fb83895d10ac44af7b9a9c3e00ca54bf49 diff --git a/gst-plugins-base-subtitles0.10/ext/vorbis/gstvorbisenc.c b/gst-plugins-base-subtitles0.10/ext/vorbis/gstvorbisenc.c new file mode 100644 index 0000000..9cdf484 --- /dev/null +++ b/gst-plugins-base-subtitles0.10/ext/vorbis/gstvorbisenc.c @@ -0,0 +1,1427 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-vorbisenc + * @see_also: vorbisdec, oggmux + * + * This element encodes raw float audio into a Vorbis stream. + * Vorbis is a royalty-free + * audio codec maintained by the Xiph.org + * Foundation. + * + * + * Example pipelines + * |[ + * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! vorbisenc ! oggmux ! filesink location=sine.ogg + * ]| Encode a test sine signal to Ogg/Vorbis. Note that the resulting file + * will be really small because a sine signal compresses very well. + * |[ + * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg + * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis. + * + * + * Last reviewed on 2006-03-01 (0.10.4) + */ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include + +#include +#include +#include +#include +#include "gstvorbisenc.h" + +#include "gstvorbiscommon.h" + +GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug); +#define GST_CAT_DEFAULT vorbisenc_debug + +static GstStaticPadTemplate vorbis_enc_sink_factory = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-float, " + "rate = (int) [ 1, 200000 ], " + "channels = (int) [ 1, 256 ], " "endianness = (int) BYTE_ORDER, " + "width = (int) 32") + ); + +static GstStaticPadTemplate vorbis_enc_src_factory = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-vorbis") + ); + +enum +{ + ARG_0, + ARG_MAX_BITRATE, + ARG_BITRATE, + ARG_MIN_BITRATE, + ARG_QUALITY, + ARG_MANAGED, + ARG_LAST_MESSAGE +}; + +static GstFlowReturn gst_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc); + +/* this function takes into account the granulepos_offset and the subgranule + * time offset */ +static GstClockTime +granulepos_to_timestamp_offset (GstVorbisEnc * vorbisenc, + ogg_int64_t granulepos) +{ + if (granulepos >= 0) + return gst_util_uint64_scale ((guint64) granulepos + + vorbisenc->granulepos_offset, GST_SECOND, vorbisenc->frequency) + + vorbisenc->subgranule_offset; + return GST_CLOCK_TIME_NONE; +} + +/* this function does a straight granulepos -> timestamp conversion */ +static GstClockTime +granulepos_to_timestamp (GstVorbisEnc * vorbisenc, ogg_int64_t granulepos) +{ + if (granulepos >= 0) + return gst_util_uint64_scale ((guint64) granulepos, + GST_SECOND, vorbisenc->frequency); + return GST_CLOCK_TIME_NONE; +} + +#define MAX_BITRATE_DEFAULT -1 +#define BITRATE_DEFAULT -1 +#define MIN_BITRATE_DEFAULT -1 +#define QUALITY_DEFAULT 0.3 +#define LOWEST_BITRATE 6000 /* lowest allowed for a 8 kHz stream */ +#define HIGHEST_BITRATE 250001 /* highest allowed for a 44 kHz stream */ + +static gboolean gst_vorbis_enc_sink_event (GstPad * pad, GstEvent * event); +static GstFlowReturn gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer); +static gboolean gst_vorbis_enc_setup (GstVorbisEnc * vorbisenc); + +static void gst_vorbis_enc_dispose (GObject * object); +static void gst_vorbis_enc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_vorbis_enc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static GstStateChangeReturn gst_vorbis_enc_change_state (GstElement * element, + GstStateChange transition); +static void gst_vorbis_enc_add_interfaces (GType vorbisenc_type); + +GST_BOILERPLATE_FULL (GstVorbisEnc, gst_vorbis_enc, GstElement, + GST_TYPE_ELEMENT, gst_vorbis_enc_add_interfaces); + +static void +gst_vorbis_enc_add_interfaces (GType vorbisenc_type) +{ + static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL }; + static const GInterfaceInfo preset_info = { NULL, NULL, NULL }; + + g_type_add_interface_static (vorbisenc_type, GST_TYPE_TAG_SETTER, + &tag_setter_info); + g_type_add_interface_static (vorbisenc_type, GST_TYPE_PRESET, &preset_info); +} + +static void +gst_vorbis_enc_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + GstPadTemplate *src_template, *sink_template; + + + src_template = gst_static_pad_template_get (&vorbis_enc_src_factory); + gst_element_class_add_pad_template (element_class, src_template); + + sink_template = gst_static_pad_template_get (&vorbis_enc_sink_factory); + gst_element_class_add_pad_template (element_class, sink_template); + gst_element_class_set_details_simple (element_class, + "Vorbis audio encoder", "Codec/Encoder/Audio", + "Encodes audio in Vorbis format", + "Monty , " "Wim Taymans "); +} + +static void +gst_vorbis_enc_class_init (GstVorbisEncClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + gobject_class->set_property = gst_vorbis_enc_set_property; + gobject_class->get_property = gst_vorbis_enc_get_property; + gobject_class->dispose = gst_vorbis_enc_dispose; + + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MAX_BITRATE, + g_param_spec_int ("max-bitrate", "Maximum Bitrate", + "Specify a maximum bitrate (in bps). Useful for streaming " + "applications. (-1 == disabled)", + -1, HIGHEST_BITRATE, MAX_BITRATE_DEFAULT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE, + g_param_spec_int ("bitrate", "Target Bitrate", + "Attempt to encode at a bitrate averaging this (in bps). " + "This uses the bitrate management engine, and is not recommended for most users. " + "Quality is a better alternative. (-1 == disabled)", -1, + HIGHEST_BITRATE, BITRATE_DEFAULT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MIN_BITRATE, + g_param_spec_int ("min-bitrate", "Minimum Bitrate", + "Specify a minimum bitrate (in bps). Useful for encoding for a " + "fixed-size channel. (-1 == disabled)", -1, HIGHEST_BITRATE, + MIN_BITRATE_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY, + g_param_spec_float ("quality", "Quality", + "Specify quality instead of specifying a particular bitrate.", -0.1, + 1.0, QUALITY_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MANAGED, + g_param_spec_boolean ("managed", "Managed", + "Enable bitrate management engine", FALSE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LAST_MESSAGE, + g_param_spec_string ("last-message", "last-message", + "The last status message", NULL, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_vorbis_enc_change_state); +} + +static void +gst_vorbis_enc_dispose (GObject * object) +{ + GstVorbisEnc *vorbisenc = GST_VORBISENC (object); + + if (vorbisenc->sinkcaps) { + gst_caps_unref (vorbisenc->sinkcaps); + vorbisenc->sinkcaps = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static GstCaps * +gst_vorbis_enc_generate_sink_caps (void) +{ + GstCaps *caps = gst_caps_new_empty (); + int i, c; + + gst_caps_append_structure (caps, gst_structure_new ("audio/x-raw-float", + "rate", GST_TYPE_INT_RANGE, 1, 200000, + "channels", G_TYPE_INT, 1, + "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, + NULL)); + + gst_caps_append_structure (caps, gst_structure_new ("audio/x-raw-float", + "rate", GST_TYPE_INT_RANGE, 1, 200000, + "channels", G_TYPE_INT, 2, + "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, + NULL)); + + for (i = 3; i <= 8; i++) { + GValue chanpos = { 0 }; + GValue pos = { 0 }; + GstStructure *structure; + + 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, gst_vorbis_channel_positions[i - 1][c]); + gst_value_array_append_value (&chanpos, &pos); + } + g_value_unset (&pos); + + structure = gst_structure_new ("audio/x-raw-float", + "rate", GST_TYPE_INT_RANGE, 1, 200000, + "channels", G_TYPE_INT, i, + "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); + gst_structure_set_value (structure, "channel-positions", &chanpos); + g_value_unset (&chanpos); + + gst_caps_append_structure (caps, structure); + } + + gst_caps_append_structure (caps, gst_structure_new ("audio/x-raw-float", + "rate", GST_TYPE_INT_RANGE, 1, 200000, + "channels", GST_TYPE_INT_RANGE, 9, 256, + "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, + NULL)); + + return caps; +} + +static GstCaps * +gst_vorbis_enc_sink_getcaps (GstPad * pad) +{ + GstVorbisEnc *vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad)); + + if (vorbisenc->sinkcaps == NULL) + vorbisenc->sinkcaps = gst_vorbis_enc_generate_sink_caps (); + + return gst_caps_ref (vorbisenc->sinkcaps); +} + +static gboolean +gst_vorbis_enc_sink_setcaps (GstPad * pad, GstCaps * caps) +{ + GstVorbisEnc *vorbisenc; + GstStructure *structure; + + vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad)); + vorbisenc->setup = FALSE; + + structure = gst_caps_get_structure (caps, 0); + gst_structure_get_int (structure, "channels", &vorbisenc->channels); + gst_structure_get_int (structure, "rate", &vorbisenc->frequency); + + gst_vorbis_enc_setup (vorbisenc); + + if (vorbisenc->setup) + return TRUE; + + return FALSE; +} + +static gboolean +gst_vorbis_enc_convert_src (GstPad * pad, GstFormat src_format, + gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +{ + gboolean res = TRUE; + GstVorbisEnc *vorbisenc; + gint64 avg; + + vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad)); + + if (vorbisenc->samples_in == 0 || + vorbisenc->bytes_out == 0 || vorbisenc->frequency == 0) { + gst_object_unref (vorbisenc); + return FALSE; + } + + avg = (vorbisenc->bytes_out * vorbisenc->frequency) / (vorbisenc->samples_in); + + switch (src_format) { + case GST_FORMAT_BYTES: + switch (*dest_format) { + case GST_FORMAT_TIME: + *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, avg); + break; + default: + res = FALSE; + } + break; + case GST_FORMAT_TIME: + switch (*dest_format) { + case GST_FORMAT_BYTES: + *dest_value = gst_util_uint64_scale_int (src_value, avg, GST_SECOND); + break; + default: + res = FALSE; + } + break; + default: + res = FALSE; + } + gst_object_unref (vorbisenc); + return res; +} + +static gboolean +gst_vorbis_enc_convert_sink (GstPad * pad, GstFormat src_format, + gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +{ + gboolean res = TRUE; + guint scale = 1; + gint bytes_per_sample; + GstVorbisEnc *vorbisenc; + + vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad)); + + bytes_per_sample = vorbisenc->channels * 2; + + switch (src_format) { + case GST_FORMAT_BYTES: + switch (*dest_format) { + case GST_FORMAT_DEFAULT: + if (bytes_per_sample == 0) + return FALSE; + *dest_value = src_value / bytes_per_sample; + break; + case GST_FORMAT_TIME: + { + gint byterate = bytes_per_sample * vorbisenc->frequency; + + if (byterate == 0) + return FALSE; + *dest_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, byterate); + break; + } + default: + res = FALSE; + } + break; + case GST_FORMAT_DEFAULT: + switch (*dest_format) { + case GST_FORMAT_BYTES: + *dest_value = src_value * bytes_per_sample; + break; + case GST_FORMAT_TIME: + if (vorbisenc->frequency == 0) + return FALSE; + *dest_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, + vorbisenc->frequency); + break; + default: + res = FALSE; + } + break; + case GST_FORMAT_TIME: + switch (*dest_format) { + case GST_FORMAT_BYTES: + scale = bytes_per_sample; + /* fallthrough */ + case GST_FORMAT_DEFAULT: + *dest_value = + gst_util_uint64_scale_int (src_value, + scale * vorbisenc->frequency, GST_SECOND); + break; + default: + res = FALSE; + } + break; + default: + res = FALSE; + } + gst_object_unref (vorbisenc); + return res; +} + +static gint64 +gst_vorbis_enc_get_latency (GstVorbisEnc * vorbisenc) +{ + /* FIXME, this probably depends on the bitrate and other setting but for now + * we return this value, which was obtained by totally unscientific + * measurements */ + return 58 * GST_MSECOND; +} + +static const GstQueryType * +gst_vorbis_enc_get_query_types (GstPad * pad) +{ + static const GstQueryType gst_vorbis_enc_src_query_types[] = { + GST_QUERY_POSITION, + GST_QUERY_DURATION, + GST_QUERY_CONVERT, + 0 + }; + + return gst_vorbis_enc_src_query_types; +} + +static gboolean +gst_vorbis_enc_src_query (GstPad * pad, GstQuery * query) +{ + gboolean res = TRUE; + GstVorbisEnc *vorbisenc; + GstPad *peerpad; + + vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad)); + peerpad = gst_pad_get_peer (GST_PAD (vorbisenc->sinkpad)); + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_POSITION: + { + GstFormat fmt, req_fmt; + gint64 pos, val; + + gst_query_parse_position (query, &req_fmt, NULL); + if ((res = gst_pad_query_position (peerpad, &req_fmt, &val))) { + gst_query_set_position (query, req_fmt, val); + break; + } + + fmt = GST_FORMAT_TIME; + if (!(res = gst_pad_query_position (peerpad, &fmt, &pos))) + break; + + if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) { + gst_query_set_position (query, req_fmt, val); + } + break; + } + case GST_QUERY_DURATION: + { + GstFormat fmt, req_fmt; + gint64 dur, val; + + gst_query_parse_duration (query, &req_fmt, NULL); + if ((res = gst_pad_query_duration (peerpad, &req_fmt, &val))) { + gst_query_set_duration (query, req_fmt, val); + break; + } + + fmt = GST_FORMAT_TIME; + if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur))) + break; + + if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) { + gst_query_set_duration (query, req_fmt, val); + } + break; + } + case GST_QUERY_CONVERT: + { + GstFormat src_fmt, dest_fmt; + gint64 src_val, dest_val; + + gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); + if (!(res = + gst_vorbis_enc_convert_src (pad, src_fmt, src_val, &dest_fmt, + &dest_val))) + goto error; + gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + break; + } + case GST_QUERY_LATENCY: + { + gboolean live; + GstClockTime min_latency, max_latency; + gint64 latency; + + if ((res = gst_pad_query (peerpad, query))) { + gst_query_parse_latency (query, &live, &min_latency, &max_latency); + + latency = gst_vorbis_enc_get_latency (vorbisenc); + + /* add our latency */ + min_latency += latency; + if (max_latency != -1) + max_latency += latency; + + gst_query_set_latency (query, live, min_latency, max_latency); + } + break; + } + default: + res = gst_pad_query (peerpad, query); + break; + } + +error: + gst_object_unref (peerpad); + gst_object_unref (vorbisenc); + return res; +} + +static gboolean +gst_vorbis_enc_sink_query (GstPad * pad, GstQuery * query) +{ + gboolean res = TRUE; + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_CONVERT: + { + GstFormat src_fmt, dest_fmt; + gint64 src_val, dest_val; + + gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); + if (!(res = + gst_vorbis_enc_convert_sink (pad, src_fmt, src_val, &dest_fmt, + &dest_val))) + goto error; + gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + break; + } + default: + res = gst_pad_query_default (pad, query); + break; + } + +error: + return res; +} + +static void +gst_vorbis_enc_init (GstVorbisEnc * vorbisenc, GstVorbisEncClass * klass) +{ + vorbisenc->sinkpad = + gst_pad_new_from_static_template (&vorbis_enc_sink_factory, "sink"); + gst_pad_set_event_function (vorbisenc->sinkpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_sink_event)); + gst_pad_set_chain_function (vorbisenc->sinkpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_chain)); + gst_pad_set_setcaps_function (vorbisenc->sinkpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_sink_setcaps)); + gst_pad_set_getcaps_function (vorbisenc->sinkpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_sink_getcaps)); + gst_pad_set_query_function (vorbisenc->sinkpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_sink_query)); + gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->sinkpad); + + vorbisenc->srcpad = + gst_pad_new_from_static_template (&vorbis_enc_src_factory, "src"); + gst_pad_set_query_function (vorbisenc->srcpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_src_query)); + gst_pad_set_query_type_function (vorbisenc->srcpad, + GST_DEBUG_FUNCPTR (gst_vorbis_enc_get_query_types)); + gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->srcpad); + + vorbisenc->channels = -1; + vorbisenc->frequency = -1; + + vorbisenc->managed = FALSE; + vorbisenc->max_bitrate = MAX_BITRATE_DEFAULT; + vorbisenc->bitrate = BITRATE_DEFAULT; + vorbisenc->min_bitrate = MIN_BITRATE_DEFAULT; + vorbisenc->quality = QUALITY_DEFAULT; + vorbisenc->quality_set = FALSE; + vorbisenc->last_message = NULL; +} + +static void +gst_vorbis_enc_metadata_set1 (const GstTagList * list, const gchar * tag, + gpointer vorbisenc) +{ + GstVorbisEnc *enc = GST_VORBISENC (vorbisenc); + GList *vc_list, *l; + + vc_list = gst_tag_to_vorbis_comments (list, tag); + + for (l = vc_list; l != NULL; l = l->next) { + const gchar *vc_string = (const gchar *) l->data; + gchar *key = NULL, *val = NULL; + + GST_LOG_OBJECT (vorbisenc, "vorbis comment: %s", vc_string); + if (gst_tag_parse_extended_comment (vc_string, &key, NULL, &val, TRUE)) { + vorbis_comment_add_tag (&enc->vc, key, val); + g_free (key); + g_free (val); + } + } + + g_list_foreach (vc_list, (GFunc) g_free, NULL); + g_list_free (vc_list); +} + +static void +gst_vorbis_enc_set_metadata (GstVorbisEnc * enc) +{ + GstTagList *merged_tags; + const GstTagList *user_tags; + + vorbis_comment_init (&enc->vc); + + user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)); + + GST_DEBUG_OBJECT (enc, "upstream tags = %" GST_PTR_FORMAT, enc->tags); + GST_DEBUG_OBJECT (enc, "user-set tags = %" GST_PTR_FORMAT, user_tags); + + /* gst_tag_list_merge() will handle NULL for either or both lists fine */ + merged_tags = gst_tag_list_merge (user_tags, enc->tags, + gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc))); + + if (merged_tags) { + GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags); + gst_tag_list_foreach (merged_tags, gst_vorbis_enc_metadata_set1, enc); + gst_tag_list_free (merged_tags); + } +} + +static gchar * +get_constraints_string (GstVorbisEnc * vorbisenc) +{ + gint min = vorbisenc->min_bitrate; + gint max = vorbisenc->max_bitrate; + gchar *result; + + if (min > 0 && max > 0) + result = g_strdup_printf ("(min %d bps, max %d bps)", min, max); + else if (min > 0) + result = g_strdup_printf ("(min %d bps, no max)", min); + else if (max > 0) + result = g_strdup_printf ("(no min, max %d bps)", max); + else + result = g_strdup_printf ("(no min or max)"); + + return result; +} + +static void +update_start_message (GstVorbisEnc * vorbisenc) +{ + gchar *constraints; + + g_free (vorbisenc->last_message); + + if (vorbisenc->bitrate > 0) { + if (vorbisenc->managed) { + constraints = get_constraints_string (vorbisenc); + vorbisenc->last_message = + g_strdup_printf ("encoding at average bitrate %d bps %s", + vorbisenc->bitrate, constraints); + g_free (constraints); + } else { + vorbisenc->last_message = + g_strdup_printf + ("encoding at approximate bitrate %d bps (VBR encoding enabled)", + vorbisenc->bitrate); + } + } else { + if (vorbisenc->quality_set) { + if (vorbisenc->managed) { + constraints = get_constraints_string (vorbisenc); + vorbisenc->last_message = + g_strdup_printf + ("encoding at quality level %2.2f using constrained VBR %s", + vorbisenc->quality, constraints); + g_free (constraints); + } else { + vorbisenc->last_message = + g_strdup_printf ("encoding at quality level %2.2f", + vorbisenc->quality); + } + } else { + constraints = get_constraints_string (vorbisenc); + vorbisenc->last_message = + g_strdup_printf ("encoding using bitrate management %s", constraints); + g_free (constraints); + } + } + + g_object_notify (G_OBJECT (vorbisenc), "last_message"); +} + +static gboolean +gst_vorbis_enc_setup (GstVorbisEnc * vorbisenc) +{ + vorbisenc->setup = FALSE; + + if (vorbisenc->bitrate < 0 && vorbisenc->min_bitrate < 0 + && vorbisenc->max_bitrate < 0) { + vorbisenc->quality_set = TRUE; + } + + update_start_message (vorbisenc); + + /* choose an encoding mode */ + /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */ + vorbis_info_init (&vorbisenc->vi); + + if (vorbisenc->quality_set) { + if (vorbis_encode_setup_vbr (&vorbisenc->vi, + vorbisenc->channels, vorbisenc->frequency, + vorbisenc->quality) != 0) { + GST_ERROR_OBJECT (vorbisenc, + "vorbisenc: initialisation failed: invalid parameters for quality"); + vorbis_info_clear (&vorbisenc->vi); + return FALSE; + } + + /* do we have optional hard quality restrictions? */ + if (vorbisenc->max_bitrate > 0 || vorbisenc->min_bitrate > 0) { + struct ovectl_ratemanage_arg ai; + + vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_GET, &ai); + + ai.bitrate_hard_min = vorbisenc->min_bitrate; + ai.bitrate_hard_max = vorbisenc->max_bitrate; + ai.management_active = 1; + + vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_SET, &ai); + } + } else { + long min_bitrate, max_bitrate; + + min_bitrate = vorbisenc->min_bitrate > 0 ? vorbisenc->min_bitrate : -1; + max_bitrate = vorbisenc->max_bitrate > 0 ? vorbisenc->max_bitrate : -1; + + if (vorbis_encode_setup_managed (&vorbisenc->vi, + vorbisenc->channels, + vorbisenc->frequency, + max_bitrate, vorbisenc->bitrate, min_bitrate) != 0) { + GST_ERROR_OBJECT (vorbisenc, + "vorbis_encode_setup_managed " + "(c %d, rate %d, max br %ld, br %d, min br %ld) failed", + vorbisenc->channels, vorbisenc->frequency, max_bitrate, + vorbisenc->bitrate, min_bitrate); + vorbis_info_clear (&vorbisenc->vi); + return FALSE; + } + } + + if (vorbisenc->managed && vorbisenc->bitrate < 0) { + vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_AVG, NULL); + } else if (!vorbisenc->managed) { + /* Turn off management entirely (if it was turned on). */ + vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_SET, NULL); + } + vorbis_encode_setup_init (&vorbisenc->vi); + + /* set up the analysis state and auxiliary encoding storage */ + vorbis_analysis_init (&vorbisenc->vd, &vorbisenc->vi); + vorbis_block_init (&vorbisenc->vd, &vorbisenc->vb); + + vorbisenc->next_ts = 0; + + vorbisenc->setup = TRUE; + + return TRUE; +} + +static GstFlowReturn +gst_vorbis_enc_clear (GstVorbisEnc * vorbisenc) +{ + GstFlowReturn ret = GST_FLOW_OK; + + if (vorbisenc->setup) { + vorbis_analysis_wrote (&vorbisenc->vd, 0); + ret = gst_vorbis_enc_output_buffers (vorbisenc); + + vorbisenc->setup = FALSE; + } + + /* clean up and exit. vorbis_info_clear() must be called last */ + vorbis_block_clear (&vorbisenc->vb); + vorbis_dsp_clear (&vorbisenc->vd); + vorbis_info_clear (&vorbisenc->vi); + + vorbisenc->header_sent = FALSE; + + return ret; +} + +/* prepare a buffer for transmission by passing data through libvorbis */ +static GstBuffer * +gst_vorbis_enc_buffer_from_packet (GstVorbisEnc * vorbisenc, + ogg_packet * packet) +{ + GstBuffer *outbuf; + + outbuf = gst_buffer_new_and_alloc (packet->bytes); + memcpy (GST_BUFFER_DATA (outbuf), packet->packet, packet->bytes); + /* see ext/ogg/README; OFFSET_END takes "our" granulepos, OFFSET its + * time representation */ + GST_BUFFER_OFFSET_END (outbuf) = packet->granulepos + + vorbisenc->granulepos_offset; + GST_BUFFER_OFFSET (outbuf) = granulepos_to_timestamp (vorbisenc, + GST_BUFFER_OFFSET_END (outbuf)); + GST_BUFFER_TIMESTAMP (outbuf) = vorbisenc->next_ts; + + /* update the next timestamp, taking granulepos_offset and subgranule offset + * into account */ + vorbisenc->next_ts = + granulepos_to_timestamp_offset (vorbisenc, packet->granulepos) + + vorbisenc->initial_ts; + GST_BUFFER_DURATION (outbuf) = + vorbisenc->next_ts - GST_BUFFER_TIMESTAMP (outbuf); + + if (vorbisenc->next_discont) { + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); + vorbisenc->next_discont = FALSE; + } + + gst_buffer_set_caps (outbuf, vorbisenc->srccaps); + + GST_LOG_OBJECT (vorbisenc, "encoded buffer of %d bytes", + GST_BUFFER_SIZE (outbuf)); + return outbuf; +} + +/* the same as above, but different logic for setting timestamp and granulepos + * */ +static GstBuffer * +gst_vorbis_enc_buffer_from_header_packet (GstVorbisEnc * vorbisenc, + ogg_packet * packet) +{ + GstBuffer *outbuf; + + outbuf = gst_buffer_new_and_alloc (packet->bytes); + memcpy (GST_BUFFER_DATA (outbuf), packet->packet, packet->bytes); + GST_BUFFER_OFFSET (outbuf) = vorbisenc->bytes_out; + GST_BUFFER_OFFSET_END (outbuf) = 0; + GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE; + GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; + + gst_buffer_set_caps (outbuf, vorbisenc->srccaps); + + GST_DEBUG ("created header packet buffer, %d bytes", + GST_BUFFER_SIZE (outbuf)); + return outbuf; +} + +/* push out the buffer and do internal bookkeeping */ +static GstFlowReturn +gst_vorbis_enc_push_buffer (GstVorbisEnc * vorbisenc, GstBuffer * buffer) +{ + vorbisenc->bytes_out += GST_BUFFER_SIZE (buffer); + + GST_DEBUG_OBJECT (vorbisenc, + "Pushing buffer with GP %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, + GST_BUFFER_OFFSET_END (buffer), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); + return gst_pad_push (vorbisenc->srcpad, buffer); +} + +static GstFlowReturn +gst_vorbis_enc_push_packet (GstVorbisEnc * vorbisenc, ogg_packet * packet) +{ + GstBuffer *outbuf; + + outbuf = gst_vorbis_enc_buffer_from_packet (vorbisenc, packet); + return gst_vorbis_enc_push_buffer (vorbisenc, outbuf); +} + +/* Set a copy of these buffers as 'streamheader' on the caps. + * We need a copy to avoid these buffers ending up with (indirect) refs on + * themselves + */ +static GstCaps * +gst_vorbis_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1, + GstBuffer * buf2, GstBuffer * buf3) +{ + GstBuffer *buf; + GstStructure *structure; + GValue array = { 0 }; + GValue value = { 0 }; + + caps = gst_caps_make_writable (caps); + structure = gst_caps_get_structure (caps, 0); + + /* mark buffers */ + GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS); + GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS); + GST_BUFFER_FLAG_SET (buf3, GST_BUFFER_FLAG_IN_CAPS); + + /* put buffers in a fixed list */ + g_value_init (&array, GST_TYPE_ARRAY); + g_value_init (&value, GST_TYPE_BUFFER); + buf = gst_buffer_copy (buf1); + gst_value_set_buffer (&value, buf); + gst_buffer_unref (buf); + gst_value_array_append_value (&array, &value); + g_value_unset (&value); + g_value_init (&value, GST_TYPE_BUFFER); + buf = gst_buffer_copy (buf2); + gst_value_set_buffer (&value, buf); + gst_buffer_unref (buf); + gst_value_array_append_value (&array, &value); + g_value_unset (&value); + g_value_init (&value, GST_TYPE_BUFFER); + buf = gst_buffer_copy (buf3); + gst_value_set_buffer (&value, buf); + gst_buffer_unref (buf); + gst_value_array_append_value (&array, &value); + gst_structure_set_value (structure, "streamheader", &array); + g_value_unset (&value); + g_value_unset (&array); + + return caps; +} + +static gboolean +gst_vorbis_enc_sink_event (GstPad * pad, GstEvent * event) +{ + gboolean res = TRUE; + GstVorbisEnc *vorbisenc; + + vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + /* Tell the library we're at end of stream so that it can handle + * the last frame and mark end of stream in the output properly */ + GST_DEBUG_OBJECT (vorbisenc, "EOS, clearing state and sending event on"); + gst_vorbis_enc_clear (vorbisenc); + + res = gst_pad_push_event (vorbisenc->srcpad, event); + break; + case GST_EVENT_TAG: + if (vorbisenc->tags) { + GstTagList *list; + + gst_event_parse_tag (event, &list); + gst_tag_list_insert (vorbisenc->tags, list, + gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (vorbisenc))); + } else { + g_assert_not_reached (); + } + res = gst_pad_push_event (vorbisenc->srcpad, event); + break; + case GST_EVENT_NEWSEGMENT: + { + gboolean update; + gdouble rate, applied_rate; + GstFormat format; + gint64 start, stop, position; + + gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate, + &format, &start, &stop, &position); + if (format == GST_FORMAT_TIME) { + gst_segment_set_newsegment (&vorbisenc->segment, update, rate, format, + start, stop, position); + if (vorbisenc->initial_ts == GST_CLOCK_TIME_NONE) { + GST_DEBUG_OBJECT (vorbisenc, "Initial segment %" GST_SEGMENT_FORMAT, + &vorbisenc->segment); + vorbisenc->initial_ts = start; + } + } + } + /* fall through */ + default: + res = gst_pad_push_event (vorbisenc->srcpad, event); + break; + } + return res; +} + +static gboolean +gst_vorbis_enc_buffer_check_discontinuous (GstVorbisEnc * vorbisenc, + GstClockTime timestamp, GstClockTime duration) +{ + gboolean ret = FALSE; + + if (timestamp != GST_CLOCK_TIME_NONE && + vorbisenc->expected_ts != GST_CLOCK_TIME_NONE && + timestamp + duration != vorbisenc->expected_ts) { + /* It turns out that a lot of elements don't generate perfect streams due + * to rounding errors. So, we permit small errors (< 1/2 a sample) without + * causing a discont. + */ + int halfsample = GST_SECOND / vorbisenc->frequency / 2; + + if ((GstClockTimeDiff) (timestamp - vorbisenc->expected_ts) > halfsample) { + GST_DEBUG_OBJECT (vorbisenc, "Expected TS %" GST_TIME_FORMAT + ", buffer TS %" GST_TIME_FORMAT, + GST_TIME_ARGS (vorbisenc->expected_ts), GST_TIME_ARGS (timestamp)); + ret = TRUE; + } + } + + if (timestamp != GST_CLOCK_TIME_NONE && duration != GST_CLOCK_TIME_NONE) { + vorbisenc->expected_ts = timestamp + duration; + } else + vorbisenc->expected_ts = GST_CLOCK_TIME_NONE; + + return ret; +} + +static GstFlowReturn +gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer) +{ + GstVorbisEnc *vorbisenc; + GstFlowReturn ret = GST_FLOW_OK; + gfloat *data; + gulong size; + gulong i, j; + float **vorbis_buffer; + GstBuffer *buf1, *buf2, *buf3; + gboolean first = FALSE; + GstClockTime timestamp = GST_CLOCK_TIME_NONE; + GstClockTime running_time = GST_CLOCK_TIME_NONE; + + vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad)); + + if (!vorbisenc->setup) + goto not_setup; + + buffer = gst_audio_buffer_clip (buffer, &vorbisenc->segment, + vorbisenc->frequency, 4 * vorbisenc->channels); + if (buffer == NULL) { + GST_DEBUG_OBJECT (vorbisenc, "Dropping buffer, out of segment"); + return GST_FLOW_OK; + } + running_time = + gst_segment_to_running_time (&vorbisenc->segment, GST_FORMAT_TIME, + GST_BUFFER_TIMESTAMP (buffer)); + timestamp = running_time + vorbisenc->initial_ts; + GST_DEBUG_OBJECT (vorbisenc, "Initial ts is %" GST_TIME_FORMAT, + GST_TIME_ARGS (vorbisenc->initial_ts)); + if (!vorbisenc->header_sent) { + /* Vorbis streams begin with three headers; the initial header (with + most of the codec setup parameters) which is mandated by the Ogg + bitstream spec. The second header holds any comment fields. The + third header holds the bitstream codebook. We merely need to + make the headers, then pass them to libvorbis one at a time; + libvorbis handles the additional Ogg bitstream constraints */ + ogg_packet header; + ogg_packet header_comm; + ogg_packet header_code; + GstCaps *caps; + + /* first, make sure header buffers get timestamp == 0 */ + vorbisenc->next_ts = 0; + vorbisenc->granulepos_offset = 0; + vorbisenc->subgranule_offset = 0; + + GST_DEBUG_OBJECT (vorbisenc, "creating and sending header packets"); + gst_vorbis_enc_set_metadata (vorbisenc); + vorbis_analysis_headerout (&vorbisenc->vd, &vorbisenc->vc, &header, + &header_comm, &header_code); + vorbis_comment_clear (&vorbisenc->vc); + + /* create header buffers */ + buf1 = gst_vorbis_enc_buffer_from_header_packet (vorbisenc, &header); + buf2 = gst_vorbis_enc_buffer_from_header_packet (vorbisenc, &header_comm); + buf3 = gst_vorbis_enc_buffer_from_header_packet (vorbisenc, &header_code); + + /* mark and put on caps */ + vorbisenc->srccaps = gst_caps_new_simple ("audio/x-vorbis", NULL); + caps = vorbisenc->srccaps; + caps = gst_vorbis_enc_set_header_on_caps (caps, buf1, buf2, buf3); + + /* negotiate with these caps */ + GST_DEBUG ("here are the caps: %" GST_PTR_FORMAT, caps); + gst_pad_set_caps (vorbisenc->srcpad, caps); + + gst_buffer_set_caps (buf1, caps); + gst_buffer_set_caps (buf2, caps); + gst_buffer_set_caps (buf3, caps); + + /* push out buffers */ + /* push_buffer takes the reference even for failure */ + if ((ret = gst_vorbis_enc_push_buffer (vorbisenc, buf1)) != GST_FLOW_OK) + goto failed_header_push; + if ((ret = gst_vorbis_enc_push_buffer (vorbisenc, buf2)) != GST_FLOW_OK) { + buf2 = NULL; + goto failed_header_push; + } + if ((ret = gst_vorbis_enc_push_buffer (vorbisenc, buf3)) != GST_FLOW_OK) { + buf3 = NULL; + goto failed_header_push; + } + + /* now adjust starting granulepos accordingly if the buffer's timestamp is + nonzero */ + vorbisenc->next_ts = timestamp; + vorbisenc->expected_ts = timestamp; + vorbisenc->granulepos_offset = gst_util_uint64_scale + (running_time, vorbisenc->frequency, GST_SECOND); + vorbisenc->subgranule_offset = 0; + vorbisenc->subgranule_offset = + (vorbisenc->next_ts - vorbisenc->initial_ts) - + granulepos_to_timestamp_offset (vorbisenc, 0); + + vorbisenc->header_sent = TRUE; + first = TRUE; + } + + if (vorbisenc->expected_ts != GST_CLOCK_TIME_NONE && + timestamp < vorbisenc->expected_ts) { + guint64 diff = vorbisenc->expected_ts - timestamp; + guint64 diff_bytes; + + GST_WARNING_OBJECT (vorbisenc, "Buffer is older than previous " + "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT + "), cannot handle. Clipping buffer.", + GST_TIME_ARGS (timestamp), GST_TIME_ARGS (vorbisenc->expected_ts)); + + diff_bytes = + GST_CLOCK_TIME_TO_FRAMES (diff, + vorbisenc->frequency) * vorbisenc->channels * sizeof (gfloat); + if (diff_bytes >= GST_BUFFER_SIZE (buffer)) { + gst_buffer_unref (buffer); + return GST_FLOW_OK; + } + buffer = gst_buffer_make_metadata_writable (buffer); + GST_BUFFER_DATA (buffer) += diff_bytes; + GST_BUFFER_SIZE (buffer) -= diff_bytes; + + GST_BUFFER_TIMESTAMP (buffer) += diff; + if (GST_BUFFER_DURATION_IS_VALID (buffer)) + GST_BUFFER_DURATION (buffer) -= diff; + } + + if (gst_vorbis_enc_buffer_check_discontinuous (vorbisenc, timestamp, + GST_BUFFER_DURATION (buffer)) && !first) { + GST_WARNING_OBJECT (vorbisenc, + "Buffer is discontinuous, flushing encoder " + "and restarting (Discont from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT + ")", GST_TIME_ARGS (vorbisenc->next_ts), GST_TIME_ARGS (timestamp)); + /* Re-initialise encoder (there's unfortunately no API to flush it) */ + if ((ret = gst_vorbis_enc_clear (vorbisenc)) != GST_FLOW_OK) + return ret; + if (!gst_vorbis_enc_setup (vorbisenc)) + return GST_FLOW_ERROR; /* Should be impossible, we can only get here if + we successfully initialised earlier */ + + /* Now, set our granulepos offset appropriately. */ + vorbisenc->next_ts = timestamp; + /* We need to round to the nearest whole number of samples, not just do + * a truncating division here */ + vorbisenc->granulepos_offset = gst_util_uint64_scale + (running_time + GST_SECOND / vorbisenc->frequency / 2 + - vorbisenc->subgranule_offset, vorbisenc->frequency, GST_SECOND); + + vorbisenc->header_sent = TRUE; + + /* And our next output buffer must have DISCONT set on it */ + vorbisenc->next_discont = TRUE; + } + + /* Sending zero samples to libvorbis marks EOS, so we mustn't do that */ + if (GST_BUFFER_SIZE (buffer) == 0) { + gst_buffer_unref (buffer); + return GST_FLOW_OK; + } + + /* data to encode */ + data = (gfloat *) GST_BUFFER_DATA (buffer); + size = GST_BUFFER_SIZE (buffer) / (vorbisenc->channels * sizeof (float)); + + /* expose the buffer to submit data */ + vorbis_buffer = vorbis_analysis_buffer (&vorbisenc->vd, size); + + /* deinterleave samples, write the buffer data */ + for (i = 0; i < size; i++) { + for (j = 0; j < vorbisenc->channels; j++) { + vorbis_buffer[j][i] = *data++; + } + } + + /* tell the library how much we actually submitted */ + vorbis_analysis_wrote (&vorbisenc->vd, size); + + GST_LOG_OBJECT (vorbisenc, "wrote %lu samples to vorbis", size); + + vorbisenc->samples_in += size; + + gst_buffer_unref (buffer); + + ret = gst_vorbis_enc_output_buffers (vorbisenc); + + return ret; + + /* error cases */ +not_setup: + { + gst_buffer_unref (buffer); + GST_ELEMENT_ERROR (vorbisenc, CORE, NEGOTIATION, (NULL), + ("encoder not initialized (input is not audio?)")); + return GST_FLOW_UNEXPECTED; + } +failed_header_push: + { + GST_WARNING_OBJECT (vorbisenc, "Failed to push headers"); + /* buf1 is always already unreffed */ + if (buf2) + gst_buffer_unref (buf2); + if (buf3) + gst_buffer_unref (buf3); + gst_buffer_unref (buffer); + return ret; + } +} + +static GstFlowReturn +gst_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc) +{ + GstFlowReturn ret; + + /* vorbis does some data preanalysis, then divides up blocks for + more involved (potentially parallel) processing. Get a single + block for encoding now */ + while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) { + ogg_packet op; + + GST_LOG_OBJECT (vorbisenc, "analysed to a block"); + + /* analysis */ + vorbis_analysis (&vorbisenc->vb, NULL); + vorbis_bitrate_addblock (&vorbisenc->vb); + + while (vorbis_bitrate_flushpacket (&vorbisenc->vd, &op)) { + GST_LOG_OBJECT (vorbisenc, "pushing out a data packet"); + ret = gst_vorbis_enc_push_packet (vorbisenc, &op); + + if (ret != GST_FLOW_OK) + return ret; + } + } + + return GST_FLOW_OK; +} + +static void +gst_vorbis_enc_get_property (GObject * object, guint prop_id, GValue * value, + GParamSpec * pspec) +{ + GstVorbisEnc *vorbisenc; + + g_return_if_fail (GST_IS_VORBISENC (object)); + + vorbisenc = GST_VORBISENC (object); + + switch (prop_id) { + case ARG_MAX_BITRATE: + g_value_set_int (value, vorbisenc->max_bitrate); + break; + case ARG_BITRATE: + g_value_set_int (value, vorbisenc->bitrate); + break; + case ARG_MIN_BITRATE: + g_value_set_int (value, vorbisenc->min_bitrate); + break; + case ARG_QUALITY: + g_value_set_float (value, vorbisenc->quality); + break; + case ARG_MANAGED: + g_value_set_boolean (value, vorbisenc->managed); + break; + case ARG_LAST_MESSAGE: + g_value_set_string (value, vorbisenc->last_message); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_vorbis_enc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstVorbisEnc *vorbisenc; + + g_return_if_fail (GST_IS_VORBISENC (object)); + + vorbisenc = GST_VORBISENC (object); + + switch (prop_id) { + case ARG_MAX_BITRATE: + { + gboolean old_value = vorbisenc->managed; + + vorbisenc->max_bitrate = g_value_get_int (value); + if (vorbisenc->max_bitrate >= 0 + && vorbisenc->max_bitrate < LOWEST_BITRATE) { + g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE); + vorbisenc->max_bitrate = LOWEST_BITRATE; + } + if (vorbisenc->min_bitrate > 0 && vorbisenc->max_bitrate > 0) + vorbisenc->managed = TRUE; + else + vorbisenc->managed = FALSE; + + if (old_value != vorbisenc->managed) + g_object_notify (object, "managed"); + break; + } + case ARG_BITRATE: + vorbisenc->bitrate = g_value_get_int (value); + if (vorbisenc->bitrate >= 0 && vorbisenc->bitrate < LOWEST_BITRATE) { + g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE); + vorbisenc->bitrate = LOWEST_BITRATE; + } + break; + case ARG_MIN_BITRATE: + { + gboolean old_value = vorbisenc->managed; + + vorbisenc->min_bitrate = g_value_get_int (value); + if (vorbisenc->min_bitrate >= 0 + && vorbisenc->min_bitrate < LOWEST_BITRATE) { + g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE); + vorbisenc->min_bitrate = LOWEST_BITRATE; + } + if (vorbisenc->min_bitrate > 0 && vorbisenc->max_bitrate > 0) + vorbisenc->managed = TRUE; + else + vorbisenc->managed = FALSE; + + if (old_value != vorbisenc->managed) + g_object_notify (object, "managed"); + break; + } + case ARG_QUALITY: + vorbisenc->quality = g_value_get_float (value); + if (vorbisenc->quality >= 0.0) + vorbisenc->quality_set = TRUE; + else + vorbisenc->quality_set = FALSE; + break; + case ARG_MANAGED: + vorbisenc->managed = g_value_get_boolean (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstStateChangeReturn +gst_vorbis_enc_change_state (GstElement * element, GstStateChange transition) +{ + GstVorbisEnc *vorbisenc = GST_VORBISENC (element); + GstStateChangeReturn res; + + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + vorbisenc->tags = gst_tag_list_new (); + break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + vorbisenc->setup = FALSE; + vorbisenc->next_discont = FALSE; + vorbisenc->header_sent = FALSE; + gst_segment_init (&vorbisenc->segment, GST_FORMAT_TIME); + vorbisenc->initial_ts = GST_CLOCK_TIME_NONE; + break; + case GST_STATE_CHANGE_PAUSED_TO_PLAYING: + break; + default: + break; + } + + res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PLAYING_TO_PAUSED: + break; + case GST_STATE_CHANGE_PAUSED_TO_READY: + vorbis_block_clear (&vorbisenc->vb); + vorbis_dsp_clear (&vorbisenc->vd); + vorbis_info_clear (&vorbisenc->vi); + g_free (vorbisenc->last_message); + vorbisenc->last_message = NULL; + if (vorbisenc->srccaps) { + gst_caps_unref (vorbisenc->srccaps); + vorbisenc->srccaps = NULL; + } + break; + case GST_STATE_CHANGE_READY_TO_NULL: + gst_tag_list_free (vorbisenc->tags); + vorbisenc->tags = NULL; + default: + break; + } + + return res; +}