--- /dev/null
+/* GStreamer
+ * Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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.
+ */
+
+/*
+ * Based on the speexdec element.
+ */
+
+/**
+ * SECTION:element-opusdec
+ * @see_also: opusenc, oggdemux
+ *
+ * This element decodes a OPUS stream to raw integer audio.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink
+ * ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "gstopusdec.h"
+#include <string.h>
+#include <gst/tag/tag.h>
+
+GST_DEBUG_CATEGORY_STATIC (opusdec_debug);
+#define GST_CAT_DEFAULT opusdec_debug
+
+#define DEC_MAX_FRAME_SIZE 2000
+
+static GstStaticPadTemplate opus_dec_src_factory =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "rate = (int) [ 32000, 64000 ], "
+ "channels = (int) [ 1, 2 ], "
+ "endianness = (int) BYTE_ORDER, "
+ "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
+ );
+
+static GstStaticPadTemplate opus_dec_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-opus")
+ );
+
+GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstElement, GST_TYPE_ELEMENT);
+
+static gboolean opus_dec_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn opus_dec_chain (GstPad * pad, GstBuffer * buf);
+static gboolean opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps);
+static GstStateChangeReturn opus_dec_change_state (GstElement * element,
+ GstStateChange transition);
+
+static gboolean opus_dec_src_event (GstPad * pad, GstEvent * event);
+static gboolean opus_dec_src_query (GstPad * pad, GstQuery * query);
+static gboolean opus_dec_sink_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *opus_get_src_query_types (GstPad * pad);
+static const GstQueryType *opus_get_sink_query_types (GstPad * pad);
+static gboolean opus_dec_convert (GstPad * pad,
+ GstFormat src_format, gint64 src_value,
+ GstFormat * dest_format, gint64 * dest_value);
+
+static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec,
+ GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
+static GstFlowReturn opus_dec_chain_parse_header (GstOpusDec * dec,
+ GstBuffer * buf);
+#if 0
+static GstFlowReturn opus_dec_chain_parse_comments (GstOpusDec * dec,
+ GstBuffer * buf);
+#endif
+
+static void
+gst_opus_dec_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&opus_dec_src_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&opus_dec_sink_factory));
+ gst_element_class_set_details_simple (element_class, "Opus audio decoder",
+ "Codec/Decoder/Audio",
+ "decode opus streams to audio",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_opus_dec_class_init (GstOpusDecClass * klass)
+{
+ GstElementClass *gstelement_class;
+
+ gstelement_class = (GstElementClass *) klass;
+
+ gstelement_class->change_state = GST_DEBUG_FUNCPTR (opus_dec_change_state);
+
+ GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
+ "opus decoding element");
+}
+
+static void
+gst_opus_dec_reset (GstOpusDec * dec)
+{
+ gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
+ dec->granulepos = -1;
+ dec->packetno = 0;
+ dec->frame_size = 0;
+ dec->frame_samples = 960;
+ dec->frame_duration = 0;
+ if (dec->state) {
+ opus_decoder_destroy (dec->state);
+ dec->state = NULL;
+ }
+#if 0
+ if (dec->mode) {
+ opus_mode_destroy (dec->mode);
+ dec->mode = NULL;
+ }
+#endif
+
+ gst_buffer_replace (&dec->streamheader, NULL);
+ gst_buffer_replace (&dec->vorbiscomment, NULL);
+ g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (dec->extra_headers);
+ dec->extra_headers = NULL;
+
+#if 0
+ memset (&dec->header, 0, sizeof (dec->header));
+#endif
+}
+
+static void
+gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class)
+{
+ dec->sinkpad =
+ gst_pad_new_from_static_template (&opus_dec_sink_factory, "sink");
+ gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (opus_dec_chain));
+ gst_pad_set_event_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (opus_dec_sink_event));
+ gst_pad_set_query_type_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (opus_get_sink_query_types));
+ gst_pad_set_query_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (opus_dec_sink_query));
+ gst_pad_set_setcaps_function (dec->sinkpad,
+ GST_DEBUG_FUNCPTR (opus_dec_sink_setcaps));
+ gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
+
+ dec->srcpad = gst_pad_new_from_static_template (&opus_dec_src_factory, "src");
+ gst_pad_use_fixed_caps (dec->srcpad);
+ gst_pad_set_event_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (opus_dec_src_event));
+ gst_pad_set_query_type_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (opus_get_src_query_types));
+ gst_pad_set_query_function (dec->srcpad,
+ GST_DEBUG_FUNCPTR (opus_dec_src_query));
+ gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
+
+ dec->sample_rate = 48000;
+ dec->n_channels = 2;
+
+ gst_opus_dec_reset (dec);
+}
+
+static gboolean
+opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+ gboolean ret = TRUE;
+ GstStructure *s;
+ const GValue *streamheader;
+
+ s = gst_caps_get_structure (caps, 0);
+ if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
+ G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
+ gst_value_array_get_size (streamheader) >= 2) {
+ const GValue *header;
+ GstBuffer *buf;
+ GstFlowReturn res = GST_FLOW_OK;
+
+ header = gst_value_array_get_value (streamheader, 0);
+ if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
+ buf = gst_value_get_buffer (header);
+ res = opus_dec_chain_parse_header (dec, buf);
+ if (res != GST_FLOW_OK)
+ goto done;
+ gst_buffer_replace (&dec->streamheader, buf);
+ }
+#if 0
+ vorbiscomment = gst_value_array_get_value (streamheader, 1);
+ if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
+ buf = gst_value_get_buffer (vorbiscomment);
+ res = opus_dec_chain_parse_comments (dec, buf);
+ if (res != GST_FLOW_OK)
+ goto done;
+ gst_buffer_replace (&dec->vorbiscomment, buf);
+ }
+#endif
+
+ g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (dec->extra_headers);
+ dec->extra_headers = NULL;
+
+ if (gst_value_array_get_size (streamheader) > 2) {
+ gint i, n;
+
+ n = gst_value_array_get_size (streamheader);
+ for (i = 2; i < n; i++) {
+ header = gst_value_array_get_value (streamheader, i);
+ buf = gst_value_get_buffer (header);
+ dec->extra_headers =
+ g_list_prepend (dec->extra_headers, gst_buffer_ref (buf));
+ }
+ }
+ }
+
+done:
+ gst_object_unref (dec);
+ return ret;
+}
+
+static gboolean
+opus_dec_convert (GstPad * pad,
+ GstFormat src_format, gint64 src_value,
+ GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = TRUE;
+ GstOpusDec *dec;
+ guint64 scale = 1;
+
+ dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ if (dec->packetno < 1) {
+ res = FALSE;
+ goto cleanup;
+ }
+
+ if (src_format == *dest_format) {
+ *dest_value = src_value;
+ res = TRUE;
+ goto cleanup;
+ }
+
+ if (pad == dec->sinkpad &&
+ (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) {
+ res = FALSE;
+ goto cleanup;
+ }
+
+ switch (src_format) {
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ scale = sizeof (gint16) * dec->n_channels;
+ case GST_FORMAT_DEFAULT:
+ *dest_value =
+ gst_util_uint64_scale_int (scale * src_value,
+ dec->sample_rate, GST_SECOND);
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = src_value * sizeof (gint16) * dec->n_channels;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value =
+ gst_util_uint64_scale_int (src_value, GST_SECOND,
+ dec->sample_rate);
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_DEFAULT:
+ *dest_value = src_value / (sizeof (gint16) * dec->n_channels);
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
+ dec->sample_rate * sizeof (gint16) * dec->n_channels);
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+
+cleanup:
+ gst_object_unref (dec);
+ return res;
+}
+
+static const GstQueryType *
+opus_get_sink_query_types (GstPad * pad)
+{
+ static const GstQueryType opus_dec_sink_query_types[] = {
+ GST_QUERY_CONVERT,
+ 0
+ };
+
+ return opus_dec_sink_query_types;
+}
+
+static gboolean
+opus_dec_sink_query (GstPad * pad, GstQuery * query)
+{
+ GstOpusDec *dec;
+ gboolean res;
+
+ dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ 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);
+ res = opus_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val);
+ if (res) {
+ gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (dec);
+ return res;
+}
+
+static const GstQueryType *
+opus_get_src_query_types (GstPad * pad)
+{
+ static const GstQueryType opus_dec_src_query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ 0
+ };
+
+ return opus_dec_src_query_types;
+}
+
+static gboolean
+opus_dec_src_query (GstPad * pad, GstQuery * query)
+{
+ GstOpusDec *dec;
+ gboolean res = FALSE;
+
+ dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_POSITION:{
+ GstSegment segment;
+ GstFormat format;
+ gint64 cur;
+
+ gst_query_parse_position (query, &format, NULL);
+
+ GST_PAD_STREAM_LOCK (dec->sinkpad);
+ segment = dec->segment;
+ GST_PAD_STREAM_UNLOCK (dec->sinkpad);
+
+ if (segment.format != GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (dec, "segment not initialised yet");
+ break;
+ }
+
+ if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
+ segment.last_stop, &format, &cur))) {
+ gst_query_set_position (query, format, cur);
+ }
+ break;
+ }
+ case GST_QUERY_DURATION:{
+ GstFormat format = GST_FORMAT_TIME;
+ gint64 dur;
+
+ /* get duration from demuxer */
+ if (!gst_pad_query_peer_duration (dec->sinkpad, &format, &dur))
+ break;
+
+ gst_query_parse_duration (query, &format, NULL);
+
+ /* and convert it into the requested format */
+ if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
+ dur, &format, &dur))) {
+ gst_query_set_duration (query, format, dur);
+ }
+ break;
+ }
+ default:
+ res = gst_pad_query_default (pad, query);
+ break;
+ }
+
+ gst_object_unref (dec);
+ return res;
+}
+
+static gboolean
+opus_dec_src_event (GstPad * pad, GstEvent * event)
+{
+ gboolean res = FALSE;
+ GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEEK:{
+ GstFormat format, tformat;
+ gdouble rate;
+ GstEvent *real_seek;
+ GstSeekFlags flags;
+ GstSeekType cur_type, stop_type;
+ gint64 cur, stop;
+ gint64 tcur, tstop;
+
+ gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+ &stop_type, &stop);
+
+ /* we have to ask our peer to seek to time here as we know
+ * nothing about how to generate a granulepos from the src
+ * formats or anything.
+ *
+ * First bring the requested format to time
+ */
+ tformat = GST_FORMAT_TIME;
+ if (!(res = opus_dec_convert (pad, format, cur, &tformat, &tcur)))
+ break;
+ if (!(res = opus_dec_convert (pad, format, stop, &tformat, &tstop)))
+ break;
+
+ /* then seek with time on the peer */
+ real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
+ flags, cur_type, tcur, stop_type, tstop);
+
+ GST_LOG_OBJECT (dec, "seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (tcur));
+
+ res = gst_pad_push_event (dec->sinkpad, real_seek);
+ gst_event_unref (event);
+ break;
+ }
+ default:
+ res = gst_pad_event_default (pad, event);
+ break;
+ }
+
+ gst_object_unref (dec);
+ return res;
+}
+
+static gboolean
+opus_dec_sink_event (GstPad * pad, GstEvent * event)
+{
+ GstOpusDec *dec;
+ gboolean ret = FALSE;
+
+ dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_NEWSEGMENT:{
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ if (format != GST_FORMAT_TIME)
+ goto newseg_wrong_format;
+
+ if (rate <= 0.0)
+ goto newseg_wrong_rate;
+
+ if (update) {
+ /* time progressed without data, see if we can fill the gap with
+ * some concealment data */
+ if (dec->segment.last_stop < start) {
+ GstClockTime duration;
+
+ duration = start - dec->segment.last_stop;
+ opus_dec_chain_parse_data (dec, NULL, dec->segment.last_stop,
+ duration);
+ }
+ }
+
+ /* now configure the values */
+ gst_segment_set_newsegment_full (&dec->segment, update,
+ rate, arate, GST_FORMAT_TIME, start, stop, time);
+
+ dec->granulepos = -1;
+
+ GST_DEBUG_OBJECT (dec, "segment now: cur = %" GST_TIME_FORMAT " [%"
+ GST_TIME_FORMAT " - %" GST_TIME_FORMAT "]",
+ GST_TIME_ARGS (dec->segment.last_stop),
+ GST_TIME_ARGS (dec->segment.start),
+ GST_TIME_ARGS (dec->segment.stop));
+
+ ret = gst_pad_push_event (dec->srcpad, event);
+ break;
+ }
+ default:
+ ret = gst_pad_event_default (pad, event);
+ break;
+ }
+
+ gst_object_unref (dec);
+ return ret;
+
+ /* ERRORS */
+newseg_wrong_format:
+ {
+ GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
+ gst_object_unref (dec);
+ return FALSE;
+ }
+newseg_wrong_rate:
+ {
+ GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
+ gst_object_unref (dec);
+ return FALSE;
+ }
+}
+
+static GstFlowReturn
+opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
+{
+ GstCaps *caps;
+ //gint error = OPUS_OK;
+
+#if 0
+ dec->samples_per_frame = opus_packet_get_samples_per_frame (
+ (const unsigned char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+#endif
+
+#if 0
+ if (memcmp (dec->header.codec_id, "OPUS ", 8) != 0)
+ goto invalid_header;
+#endif
+
+#if 0
+#ifdef HAVE_OPUS_0_7
+ dec->mode =
+ opus_mode_create (dec->sample_rate, dec->header.frame_size, &error);
+#else
+ dec->mode =
+ opus_mode_create (dec->sample_rate, dec->header.nb_channels,
+ dec->header.frame_size, &error);
+#endif
+ if (!dec->mode)
+ goto mode_init_failed;
+
+ /* initialize the decoder */
+#ifdef HAVE_OPUS_0_11
+ dec->state =
+ opus_decoder_create_custom (dec->mode, dec->header.nb_channels, &error);
+#else
+#ifdef HAVE_OPUS_0_7
+ dec->state = opus_decoder_create (dec->mode, dec->header.nb_channels, &error);
+#else
+ dec->state = opus_decoder_create (dec->mode);
+#endif
+#endif
+#endif
+ dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
+ if (!dec->state)
+ goto init_failed;
+
+#if 0
+#ifdef HAVE_OPUS_0_8
+ dec->frame_size = dec->header.frame_size;
+#else
+ opus_mode_info (dec->mode, OPUS_GET_FRAME_SIZE, &dec->frame_size);
+#endif
+#endif
+
+ dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
+ GST_SECOND, dec->sample_rate);
+
+ /* set caps */
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, dec->sample_rate,
+ "channels", G_TYPE_INT, dec->n_channels,
+ "signed", G_TYPE_BOOLEAN, TRUE,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
+
+ GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
+ dec->sample_rate, dec->n_channels, dec->frame_size);
+
+ if (!gst_pad_set_caps (dec->srcpad, caps))
+ goto nego_failed;
+
+ gst_caps_unref (caps);
+ return GST_FLOW_OK;
+
+ /* ERRORS */
+#if 0
+invalid_header:
+ {
+ GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+ (NULL), ("Invalid header"));
+ return GST_FLOW_ERROR;
+ }
+mode_init_failed:
+ {
+ GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+ (NULL), ("Mode initialization failed: %d", error));
+ return GST_FLOW_ERROR;
+ }
+#endif
+init_failed:
+ {
+ GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+ (NULL), ("couldn't initialize decoder"));
+ return GST_FLOW_ERROR;
+ }
+nego_failed:
+ {
+ GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
+ (NULL), ("couldn't negotiate format"));
+ gst_caps_unref (caps);
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+}
+
+#if 0
+static GstFlowReturn
+opus_dec_chain_parse_comments (GstOpusDec * dec, GstBuffer * buf)
+{
+ GstTagList *list;
+ gchar *encoder = NULL;
+
+ list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
+
+ if (!list) {
+ GST_WARNING_OBJECT (dec, "couldn't decode comments");
+ list = gst_tag_list_new ();
+ }
+
+ if (encoder) {
+ gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+ GST_TAG_ENCODER, encoder, NULL);
+ }
+
+ gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+ GST_TAG_AUDIO_CODEC, "Opus", NULL);
+
+ if (dec->header.bytes_per_packet > 0) {
+ gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+ GST_TAG_BITRATE, (guint) dec->header.bytes_per_packet * 8, NULL);
+ }
+
+ GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
+
+ gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
+
+ g_free (encoder);
+ g_free (ver);
+
+ return GST_FLOW_OK;
+}
+#endif
+
+static GstFlowReturn
+opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
+ GstClockTime timestamp, GstClockTime duration)
+{
+ GstFlowReturn res = GST_FLOW_OK;
+ gint size;
+ guint8 *data;
+ GstBuffer *outbuf;
+ gint16 *out_data;
+ int n;
+
+ if (timestamp != -1) {
+ dec->segment.last_stop = timestamp;
+ dec->granulepos = -1;
+ }
+
+ if (dec->state == NULL) {
+ GstCaps *caps;
+
+ dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels);
+
+ /* set caps */
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, dec->sample_rate,
+ "channels", G_TYPE_INT, dec->n_channels,
+ "signed", G_TYPE_BOOLEAN, TRUE,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
+
+ GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
+ dec->sample_rate, dec->n_channels, dec->frame_size);
+
+ if (!gst_pad_set_caps (dec->srcpad, caps))
+ GST_ERROR ("nego failure");
+
+ gst_caps_unref (caps);
+ }
+
+ if (buf) {
+ data = GST_BUFFER_DATA (buf);
+ size = GST_BUFFER_SIZE (buf);
+
+ GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
+
+ /* copy timestamp */
+ } else {
+ /* concealment data, pass NULL as the bits parameters */
+ GST_DEBUG_OBJECT (dec, "creating concealment data");
+ data = NULL;
+ size = 0;
+ }
+
+ GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data));
+ GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data,
+ 48000));
+ GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data));
+
+ res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
+ GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2,
+ GST_PAD_CAPS (dec->srcpad), &outbuf);
+
+ if (res != GST_FLOW_OK) {
+ GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
+ return res;
+ }
+
+ out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
+
+ GST_LOG_OBJECT (dec, "decoding frame");
+
+ n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, TRUE);
+ if (n < 0) {
+ GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
+ return GST_FLOW_ERROR;
+ }
+
+ if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
+ timestamp = gst_util_uint64_scale_int (dec->granulepos - dec->frame_size,
+ GST_SECOND, dec->sample_rate);
+ }
+
+ GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (timestamp));
+
+ GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
+ GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
+ if (dec->discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ dec->discont = 0;
+ }
+
+ dec->segment.last_stop += dec->frame_duration;
+
+ GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
+ GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (dec->frame_duration));
+
+ res = gst_pad_push (dec->srcpad, outbuf);
+
+ if (res != GST_FLOW_OK)
+ GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
+
+ return res;
+}
+
+static GstFlowReturn
+opus_dec_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstFlowReturn res;
+ GstOpusDec *dec;
+
+ dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
+
+ if (GST_BUFFER_IS_DISCONT (buf)) {
+ dec->discont = TRUE;
+ }
+
+ res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+ GST_BUFFER_DURATION (buf));
+
+//done:
+ dec->packetno++;
+
+ gst_buffer_unref (buf);
+ gst_object_unref (dec);
+
+ return res;
+}
+
+static GstStateChangeReturn
+opus_dec_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstOpusDec *dec = GST_OPUS_DEC (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ default:
+ break;
+ }
+
+ ret = parent_class->change_state (element, transition);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ return ret;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_opus_dec_reset (dec);
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
--- /dev/null
+/* GStreamer Opus Encoder
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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.
+ */
+
+/*
+ * Based on the speexenc element
+ */
+
+/**
+ * SECTION:element-opusenc
+ * @see_also: opusdec, oggmux
+ *
+ * This element encodes raw audio to OPUS.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg
+ * ]| Encode a test sine signal to Ogg/OPUS.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <math.h>
+#include <opus/opus.h>
+
+#include <gst/gsttagsetter.h>
+#include <gst/tag/tag.h>
+#include <gst/audio/audio.h>
+#include "gstopusenc.h"
+
+GST_DEBUG_CATEGORY_STATIC (opusenc_debug);
+#define GST_CAT_DEFAULT opusenc_debug
+
+#define GST_OPUS_ENC_TYPE_BANDWIDTH (gst_opus_enc_bandwidth_get_type())
+static GType
+gst_opus_enc_bandwidth_get_type (void)
+{
+ static const GEnumValue values[] = {
+ {OPUS_BANDWIDTH_NARROWBAND, "Narrow band", "narrowband"},
+ {OPUS_BANDWIDTH_MEDIUMBAND, "Medium band", "mediumband"},
+ {OPUS_BANDWIDTH_WIDEBAND, "Wide band", "wideband"},
+ {OPUS_BANDWIDTH_SUPERWIDEBAND, "Super wide band", "superwideband"},
+ {OPUS_BANDWIDTH_FULLBAND, "Full band", "fullband"},
+ {OPUS_BANDWIDTH_AUTO, "Auto", "auto"},
+ {0, NULL, NULL}
+ };
+ static volatile GType id = 0;
+
+ if (g_once_init_enter ((gsize *) & id)) {
+ GType _id;
+
+ _id = g_enum_register_static ("GstOpusEncBandwidth", values);
+
+ g_once_init_leave ((gsize *) & id, _id);
+ }
+
+ return id;
+}
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
+ "channels = (int) [ 1, 2 ], "
+ "endianness = (int) BYTE_ORDER, "
+ "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+ );
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-opus, "
+ "rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
+ "channels = (int) [ 1, 2 ], " "frame-size = (int) [ 2, 60 ]")
+ );
+
+#define DEFAULT_AUDIO TRUE
+#define DEFAULT_BITRATE 64000
+#define DEFAULT_BANDWIDTH OPUS_BANDWIDTH_FULLBAND
+#define DEFAULT_FRAMESIZE 20
+#define DEFAULT_CBR TRUE
+#define DEFAULT_CONSTRAINED_VBR TRUE
+#define DEFAULT_COMPLEXITY 10
+#define DEFAULT_INBAND_FEC FALSE
+#define DEFAULT_DTX FALSE
+#define DEFAULT_PACKET_LOSS_PERCENT 0
+
+enum
+{
+ PROP_0,
+ PROP_AUDIO,
+ PROP_BITRATE,
+ PROP_BANDWIDTH,
+ PROP_FRAME_SIZE,
+ PROP_CBR,
+ PROP_CONSTRAINED_VBR,
+ PROP_COMPLEXITY,
+ PROP_INBAND_FEC,
+ PROP_DTX,
+ PROP_PACKET_LOSS_PERCENT
+};
+
+static void gst_opus_enc_finalize (GObject * object);
+
+static gboolean gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_opus_enc_chain (GstPad * pad, GstBuffer * buf);
+static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
+
+static void gst_opus_enc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_opus_enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static GstStateChangeReturn gst_opus_enc_change_state (GstElement * element,
+ GstStateChange transition);
+
+static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush);
+
+static void
+gst_opus_enc_setup_interfaces (GType opusenc_type)
+{
+ static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
+ const GInterfaceInfo preset_interface_info = {
+ NULL, /* interface_init */
+ NULL, /* interface_finalize */
+ NULL /* interface_data */
+ };
+
+ g_type_add_interface_static (opusenc_type, GST_TYPE_TAG_SETTER,
+ &tag_setter_info);
+ g_type_add_interface_static (opusenc_type, GST_TYPE_PRESET,
+ &preset_interface_info);
+
+ GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder");
+}
+
+GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstElement, GST_TYPE_ELEMENT,
+ gst_opus_enc_setup_interfaces);
+
+static void
+gst_opus_enc_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_factory));
+ gst_element_class_set_details_simple (element_class, "Opus audio encoder",
+ "Codec/Encoder/Audio",
+ "Encodes audio in Opus format",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_opus_enc_class_init (GstOpusEncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ gobject_class->set_property = gst_opus_enc_set_property;
+ gobject_class->get_property = gst_opus_enc_get_property;
+
+ g_object_class_install_property (gobject_class, PROP_AUDIO,
+ g_param_spec_boolean ("audio", "Audio or voice",
+ "Audio or voice", DEFAULT_AUDIO,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
+ g_param_spec_int ("bitrate", "Encoding Bit-rate",
+ "Specify an encoding bit-rate (in bps).",
+ 1, 320000, DEFAULT_BITRATE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
+ g_param_spec_enum ("bandwidth", "Band Width",
+ "Audio Band Width", GST_OPUS_ENC_TYPE_BANDWIDTH, DEFAULT_BANDWIDTH,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_FRAME_SIZE,
+ g_param_spec_int ("frame-size", "Frame Size",
+ "The duration of an audio frame, in ms", 2, 60, DEFAULT_FRAMESIZE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_CBR,
+ g_param_spec_boolean ("cbr", "Constant bit rate",
+ "Constant bit rate", DEFAULT_CBR,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_CONSTRAINED_VBR,
+ g_param_spec_boolean ("constrained-cbr", "Constrained VBR",
+ "Constrained VBR", DEFAULT_CONSTRAINED_VBR,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_COMPLEXITY,
+ g_param_spec_int ("complexity", "Complexity",
+ "Complexity", 0, 10, DEFAULT_COMPLEXITY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_INBAND_FEC,
+ g_param_spec_boolean ("inband-fec", "In-band FEC",
+ "Enable forward error correction", DEFAULT_INBAND_FEC,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_DTX,
+ g_param_spec_boolean ("dtx", "DTX",
+ "DTX", DEFAULT_DTX, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass),
+ PROP_PACKET_LOSS_PERCENT, g_param_spec_int ("packet-loss-percentage",
+ "Loss percentage", "Packet loss percentage", 0, 100,
+ DEFAULT_PACKET_LOSS_PERCENT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize);
+
+ gstelement_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_opus_enc_change_state);
+}
+
+static void
+gst_opus_enc_finalize (GObject * object)
+{
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (object);
+
+ g_object_unref (enc->adapter);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_opus_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstOpusEnc *enc;
+ GstStructure *structure;
+ GstCaps *otherpadcaps;
+
+ enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+ enc->setup = FALSE;
+ enc->frame_size = DEFAULT_FRAMESIZE;
+ otherpadcaps = gst_pad_get_allowed_caps (pad);
+
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_get_int (structure, "channels", &enc->n_channels);
+ gst_structure_get_int (structure, "rate", &enc->sample_rate);
+
+ if (otherpadcaps) {
+ if (!gst_caps_is_empty (otherpadcaps)) {
+ GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
+ gst_structure_get_int (ps, "frame-size", &enc->frame_size);
+ }
+ gst_caps_unref (otherpadcaps);
+ }
+
+ GST_ERROR_OBJECT (pad, "channels=%d rate=%d frame-size=%d",
+ enc->n_channels, enc->sample_rate, enc->frame_size);
+ switch (enc->frame_size) {
+ case 2:
+ enc->frame_samples = enc->sample_rate / 400;
+ break;
+ case 5:
+ enc->frame_samples = enc->sample_rate / 200;
+ break;
+ case 10:
+ enc->frame_samples = enc->sample_rate / 100;
+ break;
+ case 20:
+ enc->frame_samples = enc->sample_rate / 50;
+ break;
+ case 40:
+ enc->frame_samples = enc->sample_rate / 20;
+ break;
+ case 60:
+ enc->frame_samples = 3 * enc->sample_rate / 50;
+ break;
+ default:
+ return FALSE;
+ break;
+ }
+ GST_ERROR ("frame_samples %d", enc->frame_samples);
+
+ gst_opus_enc_setup (enc);
+
+ return TRUE;
+}
+
+
+static GstCaps *
+gst_opus_enc_sink_getcaps (GstPad * pad)
+{
+ GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+ GstCaps *peercaps = NULL;
+ GstOpusEnc *enc = GST_OPUS_ENC (gst_pad_get_parent_element (pad));
+
+ peercaps = gst_pad_peer_get_caps (enc->srcpad);
+
+ if (peercaps) {
+ if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) {
+ GstStructure *ps = gst_caps_get_structure (peercaps, 0);
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ gint rate, channels;
+
+ if (gst_structure_get_int (ps, "rate", &rate)) {
+ gst_structure_fixate_field_nearest_int (s, "rate", rate);
+ }
+
+ if (gst_structure_get_int (ps, "channels", &channels)) {
+ gst_structure_fixate_field_nearest_int (s, "channels", channels);
+ }
+ }
+ gst_caps_unref (peercaps);
+ }
+
+ gst_object_unref (enc);
+
+ return caps;
+}
+
+
+static gboolean
+gst_opus_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
+ GstFormat * dest_format, gint64 * dest_value)
+{
+ gboolean res = TRUE;
+ GstOpusEnc *enc;
+ gint64 avg;
+
+ enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+ if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->sample_rate == 0)
+ return FALSE;
+
+ avg = (enc->bytes_out * enc->sample_rate) / (enc->samples_in);
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = src_value * GST_SECOND / avg;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = src_value * avg / GST_SECOND;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+ return res;
+}
+
+static gboolean
+gst_opus_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;
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+ bytes_per_sample = enc->n_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 * enc->sample_rate;
+
+ if (byterate == 0)
+ return FALSE;
+ *dest_value = 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 (enc->sample_rate == 0)
+ return FALSE;
+ *dest_value = src_value * GST_SECOND / enc->sample_rate;
+ 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 = src_value * scale * enc->sample_rate / GST_SECOND;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+ return res;
+}
+
+static gint64
+gst_opus_enc_get_latency (GstOpusEnc * enc)
+{
+ return gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
+ enc->sample_rate);
+}
+
+static const GstQueryType *
+gst_opus_enc_get_query_types (GstPad * pad)
+{
+ static const GstQueryType gst_opus_enc_src_query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ GST_QUERY_CONVERT,
+ GST_QUERY_LATENCY,
+ 0
+ };
+
+ return gst_opus_enc_src_query_types;
+}
+
+static gboolean
+gst_opus_enc_src_query (GstPad * pad, GstQuery * query)
+{
+ gboolean res = TRUE;
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (gst_pad_get_parent (pad));
+
+ 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_peer_position (enc->sinkpad, &req_fmt, &val))) {
+ gst_query_set_position (query, req_fmt, val);
+ break;
+ }
+
+ fmt = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos)))
+ break;
+
+ if ((res =
+ gst_pad_query_peer_convert (enc->sinkpad, 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_peer_duration (enc->sinkpad, &req_fmt, &val))) {
+ gst_query_set_duration (query, req_fmt, val);
+ break;
+ }
+
+ fmt = GST_FORMAT_TIME;
+ if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur)))
+ break;
+
+ if ((res =
+ gst_pad_query_peer_convert (enc->sinkpad, 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_opus_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_peer_query (pad, query))) {
+ gst_query_parse_latency (query, &live, &min_latency, &max_latency);
+
+ latency = gst_opus_enc_get_latency (enc);
+
+ /* 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_peer_query (pad, query);
+ break;
+ }
+
+error:
+
+ gst_object_unref (enc);
+
+ return res;
+}
+
+static gboolean
+gst_opus_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_opus_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_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass)
+{
+ enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
+ gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
+ gst_pad_set_event_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_sinkevent));
+ gst_pad_set_chain_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_chain));
+ gst_pad_set_setcaps_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_sink_setcaps));
+ gst_pad_set_getcaps_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps));
+ gst_pad_set_query_function (enc->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_sink_query));
+
+ enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
+ gst_pad_set_query_function (enc->srcpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_src_query));
+ gst_pad_set_query_type_function (enc->srcpad,
+ GST_DEBUG_FUNCPTR (gst_opus_enc_get_query_types));
+ gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+
+ enc->n_channels = -1;
+ enc->sample_rate = -1;
+ enc->frame_samples = 0;
+
+ enc->bitrate = DEFAULT_BITRATE;
+ enc->bandwidth = DEFAULT_BANDWIDTH;
+ enc->frame_size = DEFAULT_FRAMESIZE;
+ enc->cbr = DEFAULT_CBR;
+ enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR;
+ enc->complexity = DEFAULT_COMPLEXITY;
+ enc->inband_fec = DEFAULT_INBAND_FEC;
+ enc->dtx = DEFAULT_DTX;
+ enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT;
+
+ enc->setup = FALSE;
+ enc->header_sent = FALSE;
+
+ enc->adapter = gst_adapter_new ();
+}
+
+#if 0
+static GstBuffer *
+gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc)
+{
+ const GstTagList *tags;
+ GstTagList *empty_tags = NULL;
+ GstBuffer *comments = NULL;
+
+ tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
+
+ GST_DEBUG_OBJECT (enc, "tags = %" GST_PTR_FORMAT, tags);
+
+ if (tags == NULL) {
+ /* FIXME: better fix chain of callers to not write metadata at all,
+ * if there is none */
+ empty_tags = gst_tag_list_new ();
+ tags = empty_tags;
+ }
+ comments = gst_tag_list_to_vorbiscomment_buffer (tags, NULL,
+ 0, "Encoded with GStreamer Opusenc");
+
+ GST_BUFFER_OFFSET (comments) = enc->bytes_out;
+ GST_BUFFER_OFFSET_END (comments) = 0;
+
+ if (empty_tags)
+ gst_tag_list_free (empty_tags);
+
+ return comments;
+}
+#endif
+
+static gboolean
+gst_opus_enc_setup (GstOpusEnc * enc)
+{
+ //gint error = OPUS_OK;
+
+ enc->setup = FALSE;
+
+#if 0
+#ifdef HAVE_OPUS_0_7
+ enc->mode = opus_mode_create (enc->rate, enc->frame_size, &error);
+#else
+ enc->mode =
+ opus_mode_create (enc->rate, enc->n_channels, enc->frame_size, &error);
+#endif
+ if (!enc->mode)
+ goto mode_initialization_failed;
+
+#ifdef HAVE_OPUS_0_11
+ opus_header_init (&enc->header, enc->mode, enc->frame_size, enc->n_channels);
+#else
+#ifdef HAVE_OPUS_0_7
+ opus_header_init (&enc->header, enc->mode, enc->n_channels);
+#else
+ opus_header_init (&enc->header, enc->mode);
+#endif
+#endif
+ enc->header.nb_channels = enc->n_channels;
+
+#ifdef HAVE_OPUS_0_8
+ enc->frame_size = enc->header.frame_size;
+#else
+ opus_mode_info (enc->mode, OPUS_GET_FRAME_SIZE, &enc->frame_size);
+#endif
+#endif
+
+#if 0
+#ifdef HAVE_OPUS_0_11
+ enc->state = opus_encoder_create_custom (enc->mode, enc->n_channels, &error);
+#else
+#ifdef HAVE_OPUS_0_7
+ enc->state = opus_encoder_create (enc->mode, enc->n_channels, &error);
+#else
+ enc->state = opus_encoder_create (enc->mode);
+#endif
+#endif
+#endif
+ enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels,
+ enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP);
+ if (!enc->state)
+ goto encoder_creation_failed;
+
+ opus_encoder_ctl (enc->state, OPUS_SET_BITRATE (enc->bitrate), 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_BANDWIDTH (enc->bandwidth), 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_VBR_FLAG (!enc->cbr), 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_VBR_CONSTRAINT (enc->constrained_vbr),
+ 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_COMPLEXITY (enc->complexity), 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_INBAND_FEC_FLAG (enc->inband_fec), 0);
+ opus_encoder_ctl (enc->state, OPUS_SET_DTX_FLAG (enc->dtx), 0);
+ opus_encoder_ctl (enc->state,
+ OPUS_SET_PACKET_LOSS_PERC (enc->packet_loss_percentage), 0);
+
+ GST_LOG_OBJECT (enc, "we have frame size %d", enc->frame_size);
+
+ enc->setup = TRUE;
+
+ return TRUE;
+
+#if 0
+mode_initialization_failed:
+ GST_ERROR_OBJECT (enc, "Mode initialization failed: %d", error);
+ return FALSE;
+#endif
+
+encoder_creation_failed:
+ GST_ERROR_OBJECT (enc, "Encoder creation failed");
+ return FALSE;
+}
+
+
+/* push out the buffer and do internal bookkeeping */
+static GstFlowReturn
+gst_opus_enc_push_buffer (GstOpusEnc * enc, GstBuffer * buffer)
+{
+ guint size;
+
+ size = GST_BUFFER_SIZE (buffer);
+
+ enc->bytes_out += size;
+
+ GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
+
+ return gst_pad_push (enc->srcpad, buffer);
+}
+
+#if 0
+static GstCaps *
+gst_opus_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1,
+ GstBuffer * buf2)
+{
+ GstStructure *structure = NULL;
+ GstBuffer *buf;
+ GValue array = { 0 };
+ GValue value = { 0 };
+
+ caps = gst_caps_make_writable (caps);
+ structure = gst_caps_get_structure (caps, 0);
+
+ g_assert (gst_buffer_is_metadata_writable (buf1));
+ g_assert (gst_buffer_is_metadata_writable (buf2));
+
+ /* mark buffers */
+ GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf2, 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);
+ gst_structure_set_value (structure, "streamheader", &array);
+ g_value_unset (&value);
+ g_value_unset (&array);
+
+ return caps;
+}
+#endif
+
+
+static gboolean
+gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event)
+{
+ gboolean res = TRUE;
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_EOS:
+ gst_opus_enc_encode (enc, TRUE);
+ res = gst_pad_event_default (pad, event);
+ break;
+ case GST_EVENT_TAG:
+ {
+ GstTagList *list;
+ GstTagSetter *setter = GST_TAG_SETTER (enc);
+ const GstTagMergeMode mode = gst_tag_setter_get_tag_merge_mode (setter);
+
+ gst_event_parse_tag (event, &list);
+ gst_tag_setter_merge_tags (setter, list, mode);
+ res = gst_pad_event_default (pad, event);
+ break;
+ }
+ default:
+ res = gst_pad_event_default (pad, event);
+ break;
+ }
+
+ gst_object_unref (enc);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush)
+{
+
+ GstFlowReturn ret = GST_FLOW_OK;
+ gint bytes = enc->frame_samples * 2 * enc->n_channels;
+ gint bytes_per_packet;
+
+ bytes_per_packet =
+ (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8;
+
+ if (flush && gst_adapter_available (enc->adapter) % bytes != 0) {
+ guint diff = gst_adapter_available (enc->adapter) % bytes;
+ GstBuffer *buf = gst_buffer_new_and_alloc (diff);
+
+ memset (GST_BUFFER_DATA (buf), 0, diff);
+ gst_adapter_push (enc->adapter, buf);
+ }
+
+
+ while (gst_adapter_available (enc->adapter) >= bytes) {
+ gint16 *data;
+ gint outsize;
+ GstBuffer *outbuf;
+
+ ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad,
+ GST_BUFFER_OFFSET_NONE, bytes_per_packet, GST_PAD_CAPS (enc->srcpad),
+ &outbuf);
+
+ if (GST_FLOW_OK != ret)
+ goto done;
+
+ data = (gint16 *) gst_adapter_take (enc->adapter, bytes);
+ enc->samples_in += enc->frame_samples;
+
+ GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
+ enc->frame_samples, bytes);
+
+ outsize = opus_encode (enc->state, data, enc->frame_samples,
+ GST_BUFFER_DATA (outbuf), bytes_per_packet);
+
+ g_free (data);
+
+ if (outsize < 0) {
+ GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
+ ret = GST_FLOW_ERROR;
+ goto done;
+ }
+
+ GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts +
+ gst_util_uint64_scale_int (enc->frameno_out * enc->frame_samples,
+ GST_SECOND, enc->sample_rate);
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale_int (enc->frame_samples, GST_SECOND,
+ enc->sample_rate);
+ GST_BUFFER_OFFSET (outbuf) =
+ gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND,
+ enc->sample_rate);
+
+ enc->frameno++;
+ enc->frameno_out++;
+
+ ret = gst_opus_enc_push_buffer (enc, outbuf);
+
+ if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
+ goto done;
+ }
+
+done:
+
+ return ret;
+}
+
+static GstFlowReturn
+gst_opus_enc_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstOpusEnc *enc;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ enc = GST_OPUS_ENC (GST_PAD_PARENT (pad));
+
+ if (!enc->setup)
+ goto not_setup;
+
+#if 0
+ if (!enc->header_sent) {
+ /* Opus streams begin with two 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.
+ We merely need to make the headers, then pass them to libopus
+ one at a time; libopus handles the additional Ogg bitstream
+ constraints */
+ GstBuffer *buf1, *buf2;
+ GstCaps *caps;
+ guchar data[100];
+
+ /* create header buffer */
+ opus_header_to_packet (&enc->header, data, 100);
+ buf1 = gst_opus_enc_buffer_from_data (enc, data, 100, 0);
+
+ /* create comment buffer */
+ buf2 = gst_opus_enc_create_metadata_buffer (enc);
+
+ /* mark and put on caps */
+ caps = gst_pad_get_caps (enc->srcpad);
+ caps = gst_opus_enc_set_header_on_caps (caps, buf1, buf2);
+
+ gst_caps_set_simple (caps,
+ "rate", G_TYPE_INT, enc->sample_rate,
+ "channels", G_TYPE_INT, enc->n_channels,
+ "frame-size", G_TYPE_INT, enc->frame_size, NULL);
+
+ /* negotiate with these caps */
+ GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
+ GST_LOG_OBJECT (enc, "rate=%d channels=%d frame-size=%d",
+ enc->sample_rate, enc->n_channels, enc->frame_size);
+ gst_pad_set_caps (enc->srcpad, caps);
+
+ gst_buffer_set_caps (buf1, caps);
+ gst_buffer_set_caps (buf2, caps);
+ gst_caps_unref (caps);
+
+ /* push out buffers */
+ ret = gst_opus_enc_push_buffer (enc, buf1);
+
+ if (ret != GST_FLOW_OK) {
+ gst_buffer_unref (buf2);
+ goto done;
+ }
+
+ ret = gst_opus_enc_push_buffer (enc, buf2);
+
+ if (ret != GST_FLOW_OK)
+ goto done;
+
+ enc->header_sent = TRUE;
+ }
+#endif
+
+ GST_DEBUG_OBJECT (enc, "received buffer of %u bytes", GST_BUFFER_SIZE (buf));
+
+ /* Save the timestamp of the first buffer. This will be later
+ * used as offset for all following buffers */
+ if (enc->start_ts == GST_CLOCK_TIME_NONE) {
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+ enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
+ } else {
+ enc->start_ts = 0;
+ }
+ }
+
+
+ /* Check if we have a continous stream, if not drop some samples or the buffer or
+ * insert some silence samples */
+ if (enc->next_ts != GST_CLOCK_TIME_NONE &&
+ GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
+ guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
+ guint64 diff_bytes;
+
+ GST_WARNING_OBJECT (enc, "Buffer is older than previous "
+ "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
+ "), cannot handle. Clipping buffer.",
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (enc->next_ts));
+
+ diff_bytes =
+ GST_CLOCK_TIME_TO_FRAMES (diff, enc->sample_rate) * enc->n_channels * 2;
+ if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
+ gst_buffer_unref (buf);
+ return GST_FLOW_OK;
+ }
+ buf = gst_buffer_make_metadata_writable (buf);
+ GST_BUFFER_DATA (buf) += diff_bytes;
+ GST_BUFFER_SIZE (buf) -= diff_bytes;
+
+ GST_BUFFER_TIMESTAMP (buf) += diff;
+ if (GST_BUFFER_DURATION_IS_VALID (buf))
+ GST_BUFFER_DURATION (buf) -= diff;
+ }
+
+ if (enc->next_ts != GST_CLOCK_TIME_NONE
+ && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+ guint64 max_diff =
+ gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->sample_rate);
+
+ if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
+ GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) {
+ GST_WARNING_OBJECT (enc,
+ "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
+ GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff);
+
+ gst_opus_enc_encode (enc, TRUE);
+
+ enc->frameno_out = 0;
+ enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
+ }
+ }
+
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
+ && GST_BUFFER_DURATION_IS_VALID (buf))
+ enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+ else
+ enc->next_ts = GST_CLOCK_TIME_NONE;
+
+ /* push buffer to adapter */
+ gst_adapter_push (enc->adapter, buf);
+ buf = NULL;
+
+ ret = gst_opus_enc_encode (enc, FALSE);
+
+done:
+
+ if (buf)
+ gst_buffer_unref (buf);
+
+ return ret;
+
+ /* ERRORS */
+not_setup:
+ {
+ GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
+ ("encoder not initialized (input is not audio?)"));
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto done;
+ }
+
+}
+
+
+static void
+gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (object);
+
+ switch (prop_id) {
+ case PROP_AUDIO:
+ g_value_set_boolean (value, enc->audio_or_voip);
+ break;
+ case PROP_BITRATE:
+ g_value_set_int (value, enc->bitrate);
+ break;
+ case PROP_BANDWIDTH:
+ g_value_set_int (value, enc->bandwidth);
+ break;
+ case PROP_FRAME_SIZE:
+ g_value_set_int (value, enc->frame_size);
+ break;
+ case PROP_CBR:
+ g_value_set_boolean (value, enc->cbr);
+ break;
+ case PROP_CONSTRAINED_VBR:
+ g_value_set_boolean (value, enc->constrained_vbr);
+ break;
+ case PROP_COMPLEXITY:
+ g_value_set_int (value, enc->complexity);
+ break;
+ case PROP_INBAND_FEC:
+ g_value_set_boolean (value, enc->inband_fec);
+ break;
+ case PROP_DTX:
+ g_value_set_boolean (value, enc->dtx);
+ break;
+ case PROP_PACKET_LOSS_PERCENT:
+ g_value_set_int (value, enc->packet_loss_percentage);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_opus_enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstOpusEnc *enc;
+
+ enc = GST_OPUS_ENC (object);
+
+ switch (prop_id) {
+ case PROP_AUDIO:
+ enc->audio_or_voip = g_value_get_boolean (value);
+ break;
+ case PROP_BITRATE:
+ enc->bitrate = g_value_get_int (value);
+ break;
+ case PROP_BANDWIDTH:
+ enc->bandwidth = g_value_get_int (value);
+ break;
+ case PROP_FRAME_SIZE:
+ enc->frame_size = g_value_get_int (value);
+ break;
+ case PROP_CBR:
+ enc->cbr = g_value_get_boolean (value);
+ break;
+ case PROP_CONSTRAINED_VBR:
+ enc->constrained_vbr = g_value_get_boolean (value);
+ break;
+ case PROP_COMPLEXITY:
+ enc->complexity = g_value_get_int (value);
+ break;
+ case PROP_INBAND_FEC:
+ enc->inband_fec = g_value_get_boolean (value);
+ break;
+ case PROP_DTX:
+ enc->dtx = g_value_get_boolean (value);
+ break;
+ case PROP_PACKET_LOSS_PERCENT:
+ enc->packet_loss_percentage = g_value_get_int (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_opus_enc_change_state (GstElement * element, GstStateChange transition)
+{
+ GstOpusEnc *enc = GST_OPUS_ENC (element);
+ GstStateChangeReturn res;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ enc->frameno = 0;
+ enc->samples_in = 0;
+ enc->frameno_out = 0;
+ enc->start_ts = GST_CLOCK_TIME_NONE;
+ enc->next_ts = GST_CLOCK_TIME_NONE;
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ /* fall through */
+ default:
+ break;
+ }
+
+ res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (res == GST_STATE_CHANGE_FAILURE)
+ return res;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ enc->setup = FALSE;
+ enc->header_sent = FALSE;
+ if (enc->state) {
+ opus_encoder_destroy (enc->state);
+ enc->state = NULL;
+ }
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
+ default:
+ break;
+ }
+
+ return res;
+}
projects / platform / upstream / gst-plugins-base.git / commitdiff