From 3b6cd3d35c77f1413a8fd1d52269120c3eabfe20 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sun, 5 Jun 2011 00:54:19 -0700 Subject: [PATCH] opus: duplicate from CELT Copy the celt plugin and convert it to Opus. Mostly works. --- configure.ac | 15 + ext/Makefile.am | 8 + ext/opus/Makefile.am | 16 + ext/opus/gstopus.c | 50 ++ ext/opus/gstopusdec.c | 865 +++++++++++++++++++++++++++++ ext/opus/gstopusdec.h | 77 +++ ext/opus/gstopusenc.c | 1198 +++++++++++++++++++++++++++++++++++++++++ ext/opus/gstopusenc.h | 105 ++++ 8 files changed, 2334 insertions(+) create mode 100644 ext/opus/Makefile.am create mode 100644 ext/opus/gstopus.c create mode 100644 ext/opus/gstopusdec.c create mode 100644 ext/opus/gstopusdec.h create mode 100644 ext/opus/gstopusenc.c create mode 100644 ext/opus/gstopusenc.h diff --git a/configure.ac b/configure.ac index 7322eaa934..1aac3360a4 100644 --- a/configure.ac +++ b/configure.ac @@ -1385,6 +1385,19 @@ AG_GST_CHECK_FEATURE(OPENCV, [opencv plugins], opencv, [ AC_SUBST(OPENCV_LIBS) ]) +dnl *** Opus *** +translit(dnm, m, l) AM_CONDITIONAL(USE_OPUS, true) +AG_GST_CHECK_FEATURE(OPUS, [opus], opus, [ + PKG_CHECK_MODULES(OPUS, opus >= 0.9.4, [ + AC_DEFINE([HAVE_OPUS], 1, [Define if Opus >= 0.9.4 is installed]) + HAVE_OPUS="yes" + ], [ + HAVE_OPUS="no" + ]) + AC_SUBST(OPUS_CFLAGS) + AC_SUBST(OPUS_LIBS) +]) + dnl *** rsvg *** translit(dnm, m, l) AM_CONDITIONAL(USE_RSVG, true) AG_GST_CHECK_FEATURE(RSVG, [rsvg decoder], rsvg, [ @@ -1750,6 +1763,7 @@ AM_CONDITIONAL(USE_NEON, false) AM_CONDITIONAL(USE_OFA, false) AM_CONDITIONAL(USE_OPENAL, false) AM_CONDITIONAL(USE_OPENCV, false) +AM_CONDITIONAL(USE_OPUS, false) AM_CONDITIONAL(USE_RSVG, false) AM_CONDITIONAL(USE_TIMIDITY, false) AM_CONDITIONAL(USE_WILDMIDI, false) @@ -1994,6 +2008,7 @@ ext/neon/Makefile ext/ofa/Makefile ext/openal/Makefile ext/opencv/Makefile +ext/opus/Makefile ext/rsvg/Makefile ext/resindvd/Makefile ext/rtmp/Makefile diff --git a/ext/Makefile.am b/ext/Makefile.am index 70d4c69c31..2a6f8ec760 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -262,6 +262,12 @@ else OPENCV_DIR= endif +if USE_OPUS +OPUS_DIR=opus +else +OPUS_DIR= +endif + if USE_RSVG RSVG_DIR=rsvg else @@ -419,6 +425,7 @@ SUBDIRS=\ $(OFA_DIR) \ $(OPENAL_DIR) \ $(OPENCV_DIR) \ + $(OPUS_DIR) \ $(RSVG_DIR) \ $(SCHRO_DIR) \ $(SDL_DIR) \ @@ -471,6 +478,7 @@ DIST_SUBDIRS = \ ofa \ openal \ opencv \ + opus \ rsvg \ resindvd \ schroedinger \ diff --git a/ext/opus/Makefile.am b/ext/opus/Makefile.am new file mode 100644 index 0000000000..aa50ba96ef --- /dev/null +++ b/ext/opus/Makefile.am @@ -0,0 +1,16 @@ +plugin_LTLIBRARIES = libgstopus.la + +libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c +libgstopus_la_CFLAGS = \ + $(GST_PLUGINS_BASE_CFLAGS) \ + $(GST_CFLAGS) \ + $(OPUS_CFLAGS) +libgstopus_la_LIBADD = \ + $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \ + $(GST_BASE_LIBS) \ + $(GST_LIBS) \ + $(OPUS_LIBS) +libgstopus_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM) +libgstopus_la_LIBTOOLFLAGS = --tag=disable-static + +noinst_HEADERS = gstopusenc.h gstopusdec.h diff --git a/ext/opus/gstopus.c b/ext/opus/gstopus.c new file mode 100644 index 0000000000..65e9dcdc58 --- /dev/null +++ b/ext/opus/gstopus.c @@ -0,0 +1,50 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * Copyright (C) <2008> Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ +#ifdef HAVE_CONFIG_H +#include +#endif + +#include "gstopusdec.h" +#include "gstopusenc.h" + +#include + +static gboolean +plugin_init (GstPlugin * plugin) +{ + + if (!gst_element_register (plugin, "opusenc", GST_RANK_NONE, + GST_TYPE_OPUS_ENC)) + return FALSE; + + if (!gst_element_register (plugin, "opusdec", GST_RANK_PRIMARY, + GST_TYPE_OPUS_DEC)) + return FALSE; + + gst_tag_register_musicbrainz_tags (); + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "opus", + "OPUS plugin library", + plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c new file mode 100644 index 0000000000..47c06cec0a --- /dev/null +++ b/ext/opus/gstopusdec.c @@ -0,0 +1,865 @@ +/* GStreamer + * Copyright (C) 2004 Wim Taymans + * Copyright (C) 2006 Tim-Philipp Müller + * Copyright (C) 2008 Sebastian Dröge + * + * 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. + * + * + * Example pipelines + * |[ + * 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. + * + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include "gstopusdec.h" +#include +#include + +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 "); +} + +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; +} diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h new file mode 100644 index 0000000000..886a907532 --- /dev/null +++ b/ext/opus/gstopusdec.h @@ -0,0 +1,77 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * Copyright (C) <2008> Sebastian Dröge + * + * 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. + */ + +#ifndef __GST_OPUS_DEC_H__ +#define __GST_OPUS_DEC_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_OPUS_DEC \ + (gst_opus_dec_get_type()) +#define GST_OPUS_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OPUS_DEC,GstOpusDec)) +#define GST_OPUS_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OPUS_DEC,GstOpusDecClass)) +#define GST_IS_OPUS_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OPUS_DEC)) +#define GST_IS_OPUS_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OPUS_DEC)) + +typedef struct _GstOpusDec GstOpusDec; +typedef struct _GstOpusDecClass GstOpusDecClass; + +struct _GstOpusDec { + GstElement element; + + /* pads */ + GstPad *sinkpad; + GstPad *srcpad; + + OpusDecoder *state; + int frame_samples; + + gint frame_size; + GstClockTime frame_duration; + guint64 packetno; + + GstSegment segment; /* STREAM LOCK */ + gint64 granulepos; /* -1 = needs to be set from current time */ + gboolean discont; + + GstBuffer *streamheader; + GstBuffer *vorbiscomment; + GList *extra_headers; + + int sample_rate; + int n_channels; +}; + +struct _GstOpusDecClass { + GstElementClass parent_class; +}; + +GType gst_opus_dec_get_type (void); + +G_END_DECLS + +#endif /* __GST_OPUS_DEC_H__ */ diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c new file mode 100644 index 0000000000..db57ff75d1 --- /dev/null +++ b/ext/opus/gstopusenc.c @@ -0,0 +1,1198 @@ +/* GStreamer Opus Encoder + * Copyright (C) <1999> Erik Walthinsen + * Copyright (C) <2008> Sebastian Dröge + * + * 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. + * + * + * Example pipelines + * |[ + * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg + * ]| Encode a test sine signal to Ogg/OPUS. + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include +#include +#include +#include +#include + +#include +#include +#include +#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 "); +} + +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; +} diff --git a/ext/opus/gstopusenc.h b/ext/opus/gstopusenc.h new file mode 100644 index 0000000000..5cb54598af --- /dev/null +++ b/ext/opus/gstopusenc.h @@ -0,0 +1,105 @@ +/* GStreamer Opus Encoder + * Copyright (C) <1999> Erik Walthinsen + * Copyright (C) <2008> Sebastian Dröge + * + * 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. + */ + + +#ifndef __GST_OPUS_ENC_H__ +#define __GST_OPUS_ENC_H__ + + +#include +#include + +#include + +G_BEGIN_DECLS + +#define GST_TYPE_OPUS_ENC \ + (gst_opus_enc_get_type()) +#define GST_OPUS_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OPUS_ENC,GstOpusEnc)) +#define GST_OPUS_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OPUS_ENC,GstOpusEncClass)) +#define GST_IS_OPUS_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OPUS_ENC)) +#define GST_IS_OPUS_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OPUS_ENC)) + +#define MAX_FRAME_SIZE 2000*2 +#define MAX_FRAME_BYTES 2000 + +typedef struct _GstOpusEnc GstOpusEnc; +typedef struct _GstOpusEncClass GstOpusEncClass; + +struct _GstOpusEnc { + GstElement element; + + /* pads */ + GstPad *sinkpad; + GstPad *srcpad; + + //OpusHeader header; + //OpusMode *mode; + OpusEncoder *state; + GstAdapter *adapter; + + /* properties */ + gboolean audio_or_voip; + gint bitrate; + gint bandwidth; + gint frame_size; + gboolean cbr; + gboolean constrained_vbr; + gint complexity; + gboolean inband_fec; + gboolean dtx; + gint packet_loss_percentage; + + int frame_samples; + + gint n_channels; + gint sample_rate; + + gboolean setup; + gboolean header_sent; + gboolean eos; + + guint64 samples_in; + guint64 bytes_out; + + guint64 frameno; + guint64 frameno_out; + + GstClockTime start_ts; + GstClockTime next_ts; + guint64 granulepos_offset; +}; + +struct _GstOpusEncClass { + GstElementClass parent_class; + + /* signals */ + void (*frame_encoded) (GstElement *element); +}; + +GType gst_opus_enc_get_type (void); + +G_END_DECLS + +#endif /* __GST_OPUS_ENC_H__ */