From 715f9ce62a12d128754d2cf47f90a024b537e320 Mon Sep 17 00:00:00 2001 From: HyungKyu Song Date: Sat, 16 Feb 2013 00:14:42 +0900 Subject: Tizen 2.0 Release --- ext/speex/Makefile.am | 18 + ext/speex/Makefile.in | 834 +++++++++++++++++++++++++++++++++++++++++++++++ ext/speex/gstspeex.c | 49 +++ ext/speex/gstspeexdec.c | 549 +++++++++++++++++++++++++++++++ ext/speex/gstspeexdec.h | 80 +++++ ext/speex/gstspeexenc.c | 850 ++++++++++++++++++++++++++++++++++++++++++++++++ ext/speex/gstspeexenc.h | 102 ++++++ 7 files changed, 2482 insertions(+) create mode 100644 ext/speex/Makefile.am create mode 100644 ext/speex/Makefile.in create mode 100644 ext/speex/gstspeex.c create mode 100644 ext/speex/gstspeexdec.c create mode 100644 ext/speex/gstspeexdec.h create mode 100644 ext/speex/gstspeexenc.c create mode 100644 ext/speex/gstspeexenc.h (limited to 'ext/speex') diff --git a/ext/speex/Makefile.am b/ext/speex/Makefile.am new file mode 100644 index 0000000..fe55237 --- /dev/null +++ b/ext/speex/Makefile.am @@ -0,0 +1,18 @@ +plugin_LTLIBRARIES = libgstspeex.la + +libgstspeex_la_SOURCES = gstspeex.c gstspeexdec.c gstspeexenc.c +libgstspeex_la_CFLAGS = -DGST_USE_UNSTABLE_API \ + $(GST_PLUGINS_BASE_CFLAGS) \ + $(GST_BASE_CFLAGS) \ + $(GST_CFLAGS) \ + $(SPEEX_CFLAGS) +libgstspeex_la_LIBADD = \ + $(GST_PLUGINS_BASE_LIBS) \ + -lgsttag-$(GST_MAJORMINOR) -lgstaudio-$(GST_MAJORMINOR) \ + $(GST_BASE_LIBS) \ + $(GST_LIBS) \ + $(SPEEX_LIBS) +libgstspeex_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(LIBM) +libgstspeex_la_LIBTOOLFLAGS = --tag=disable-static + +noinst_HEADERS = gstspeexenc.h gstspeexdec.h diff --git a/ext/speex/Makefile.in b/ext/speex/Makefile.in new file mode 100644 index 0000000..8f5bcef --- /dev/null +++ b/ext/speex/Makefile.in @@ -0,0 +1,834 @@ +# Makefile.in generated by automake 1.11.3 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002, +# 2003, 2004, 2005, 2006, 2007, 2008, 2009, 2010, 2011 Free Software 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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 "gstspeexdec.h" +#include "gstspeexenc.h" + +#include + +static gboolean +plugin_init (GstPlugin * plugin) +{ + + if (!gst_element_register (plugin, "speexenc", GST_RANK_PRIMARY, + GST_TYPE_SPEEX_ENC)) + return FALSE; + + if (!gst_element_register (plugin, "speexdec", GST_RANK_PRIMARY, + GST_TYPE_SPEEX_DEC)) + return FALSE; + + gst_tag_register_musicbrainz_tags (); + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "speex", + "Speex plugin library", + plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c new file mode 100644 index 0000000..e1d80b6 --- /dev/null +++ b/ext/speex/gstspeexdec.c @@ -0,0 +1,549 @@ +/* GStreamer + * Copyright (C) 2004 Wim Taymans + * Copyright (C) 2006 Tim-Philipp Müller + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-speexdec + * @see_also: speexenc, oggdemux + * + * This element decodes a Speex stream to raw integer audio. + * Speex is a royalty-free + * audio codec maintained by the Xiph.org + * Foundation. + * + * + * Example pipelines + * |[ + * gst-launch -v filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink + * ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the + * documentation of speexenc. + * + * + * Last reviewed on 2006-04-05 (0.10.2) + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include "gstspeexdec.h" +#include +#include +#include + +GST_DEBUG_CATEGORY_STATIC (speexdec_debug); +#define GST_CAT_DEFAULT speexdec_debug + +#define DEFAULT_ENH TRUE + +enum +{ + ARG_0, + ARG_ENH +}; + +static GstStaticPadTemplate speex_dec_src_factory = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "rate = (int) [ 6000, 48000 ], " + "channels = (int) [ 1, 2 ], " + "endianness = (int) BYTE_ORDER, " + "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16") + ); + +static GstStaticPadTemplate speex_dec_sink_factory = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-speex") + ); + +GST_BOILERPLATE (GstSpeexDec, gst_speex_dec, GstAudioDecoder, + GST_TYPE_AUDIO_DECODER); + + +static gboolean gst_speex_dec_start (GstAudioDecoder * dec); +static gboolean gst_speex_dec_stop (GstAudioDecoder * dec); +static gboolean gst_speex_dec_set_format (GstAudioDecoder * bdec, + GstCaps * caps); +static GstFlowReturn gst_speex_dec_handle_frame (GstAudioDecoder * dec, + GstBuffer * buffer); + +static void gst_speex_dec_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_speex_dec_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); + +static void +gst_speex_dec_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_static_pad_template (element_class, + &speex_dec_src_factory); + gst_element_class_add_static_pad_template (element_class, + &speex_dec_sink_factory); + gst_element_class_set_details_simple (element_class, "Speex audio decoder", + "Codec/Decoder/Audio", + "decode speex streams to audio", "Wim Taymans "); +} + +static void +gst_speex_dec_class_init (GstSpeexDecClass * klass) +{ + GObjectClass *gobject_class; + GstAudioDecoderClass *base_class; + + gobject_class = (GObjectClass *) klass; + base_class = (GstAudioDecoderClass *) klass; + + gobject_class->set_property = gst_speex_dec_set_property; + gobject_class->get_property = gst_speex_dec_get_property; + + base_class->start = GST_DEBUG_FUNCPTR (gst_speex_dec_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_dec_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_dec_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_dec_handle_frame); + + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_ENH, + g_param_spec_boolean ("enh", "Enh", "Enable perceptual enhancement", + DEFAULT_ENH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + GST_DEBUG_CATEGORY_INIT (speexdec_debug, "speexdec", 0, + "speex decoding element"); +} + +static void +gst_speex_dec_reset (GstSpeexDec * dec) +{ + dec->packetno = 0; + dec->frame_size = 0; + dec->frame_duration = 0; + dec->mode = NULL; + free (dec->header); + dec->header = NULL; + speex_bits_destroy (&dec->bits); + + gst_buffer_replace (&dec->streamheader, NULL); + gst_buffer_replace (&dec->vorbiscomment, NULL); + + if (dec->stereo) { + speex_stereo_state_destroy (dec->stereo); + dec->stereo = NULL; + } + + if (dec->state) { + speex_decoder_destroy (dec->state); + dec->state = NULL; + } +} + +static void +gst_speex_dec_init (GstSpeexDec * dec, GstSpeexDecClass * g_class) +{ + dec->enh = DEFAULT_ENH; + + gst_speex_dec_reset (dec); +} + +static gboolean +gst_speex_dec_start (GstAudioDecoder * dec) +{ + GstSpeexDec *sd = GST_SPEEX_DEC (dec); + + GST_DEBUG_OBJECT (dec, "start"); + gst_speex_dec_reset (sd); + + /* we know about concealment */ + gst_audio_decoder_set_plc_aware (dec, TRUE); + + return TRUE; +} + +static gboolean +gst_speex_dec_stop (GstAudioDecoder * dec) +{ + GstSpeexDec *sd = GST_SPEEX_DEC (dec); + + GST_DEBUG_OBJECT (dec, "stop"); + gst_speex_dec_reset (sd); + + return TRUE; +} + +static GstFlowReturn +gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf) +{ + GstCaps *caps; + + /* get the header */ + dec->header = speex_packet_to_header ((char *) GST_BUFFER_DATA (buf), + GST_BUFFER_SIZE (buf)); + + if (!dec->header) + goto no_header; + + if (dec->header->mode >= SPEEX_NB_MODES || dec->header->mode < 0) + goto mode_too_old; + + dec->mode = speex_lib_get_mode (dec->header->mode); + + /* initialize the decoder */ + dec->state = speex_decoder_init (dec->mode); + if (!dec->state) + goto init_failed; + + speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh); + speex_decoder_ctl (dec->state, SPEEX_GET_FRAME_SIZE, &dec->frame_size); + + if (dec->header->nb_channels != 1) { + dec->stereo = speex_stereo_state_init (); + dec->callback.callback_id = SPEEX_INBAND_STEREO; + dec->callback.func = speex_std_stereo_request_handler; + dec->callback.data = dec->stereo; + speex_decoder_ctl (dec->state, SPEEX_SET_HANDLER, &dec->callback); + } + + speex_decoder_ctl (dec->state, SPEEX_SET_SAMPLING_RATE, &dec->header->rate); + + dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size, + GST_SECOND, dec->header->rate); + + speex_bits_init (&dec->bits); + + /* set caps */ + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, dec->header->rate, + "channels", G_TYPE_INT, dec->header->nb_channels, + "signed", G_TYPE_BOOLEAN, TRUE, + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL); + + if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps)) + goto nego_failed; + + gst_caps_unref (caps); + return GST_FLOW_OK; + + /* ERRORS */ +no_header: + { + GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE, + (NULL), ("couldn't read header")); + return GST_FLOW_ERROR; + } +mode_too_old: + { + GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE, + (NULL), + ("Mode number %d does not (yet/any longer) exist in this version", + dec->header->mode)); + return GST_FLOW_ERROR; + } +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; + } +} + +static GstFlowReturn +gst_speex_dec_parse_comments (GstSpeexDec * dec, GstBuffer * buf) +{ + GstTagList *list; + gchar *ver, *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, "Speex", NULL); + + ver = g_strndup (dec->header->speex_version, SPEEX_HEADER_VERSION_LENGTH); + g_strstrip (ver); + + if (ver != NULL && *ver != '\0') { + gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, + GST_TAG_ENCODER_VERSION, ver, NULL); + } + + if (dec->header->bitrate > 0) { + gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, + GST_TAG_BITRATE, (guint) dec->header->bitrate, NULL); + } + + GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list); + + gst_element_found_tags_for_pad (GST_ELEMENT (dec), + GST_AUDIO_DECODER_SRC_PAD (dec), list); + + g_free (encoder); + g_free (ver); + + return GST_FLOW_OK; +} + +static gboolean +gst_speex_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) +{ + GstSpeexDec *dec = GST_SPEEX_DEC (bdec); + 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, *vorbiscomment; + 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 = gst_speex_dec_parse_header (dec, buf); + if (res != GST_FLOW_OK) + goto done; + gst_buffer_replace (&dec->streamheader, buf); + } + + vorbiscomment = gst_value_array_get_value (streamheader, 1); + if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) { + buf = gst_value_get_buffer (vorbiscomment); + res = gst_speex_dec_parse_comments (dec, buf); + if (res != GST_FLOW_OK) + goto done; + gst_buffer_replace (&dec->vorbiscomment, buf); + } + } + +done: + return ret; +} + +static GstFlowReturn +gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf) +{ + GstFlowReturn res = GST_FLOW_OK; + gint i, fpp; + guint size; + guint8 *data; + SpeexBits *bits; + + if (!dec->frame_duration) + goto not_negotiated; + + if (G_LIKELY (GST_BUFFER_SIZE (buf))) { + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + + /* send data to the bitstream */ + speex_bits_read_from (&dec->bits, (char *) data, size); + + fpp = dec->header->frames_per_packet; + bits = &dec->bits; + + GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d, %d bits", + size, fpp, speex_bits_remaining (bits)); + } else { + /* FIXME ? actually consider how much concealment is needed */ + /* concealment data, pass NULL as the bits parameters */ + GST_DEBUG_OBJECT (dec, "creating concealment data"); + fpp = dec->header->frames_per_packet; + bits = NULL; + } + + /* now decode each frame, catering for unknown number of them (e.g. rtp) */ + for (i = 0; i < fpp; i++) { + GstBuffer *outbuf; + gint16 *out_data; + gint ret; + + GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp, + bits ? speex_bits_remaining (bits) : -1); + + res = + gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), + GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2, + GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &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); + + ret = speex_decode_int (dec->state, bits, out_data); + if (ret == -1) { + /* uh? end of stream */ + if (fpp == 0 && speex_bits_remaining (bits) < 8) { + /* if we did not know how many frames to expect, then we get this + at the end if there are leftover bits to pad to the next byte */ + GST_DEBUG_OBJECT (dec, "Discarding leftover bits"); + } else { + GST_WARNING_OBJECT (dec, "Unexpected end of stream found"); + } + gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1); + gst_buffer_unref (outbuf); + } else if (ret == -2) { + GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?"); + gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1); + gst_buffer_unref (outbuf); + } + + if (bits && speex_bits_remaining (bits) < 0) { + GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?"); + gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1); + gst_buffer_unref (outbuf); + } + if (dec->header->nb_channels == 2) + speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo); + + res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1); + + if (res != GST_FLOW_OK) { + GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); + break; + } + } + + return res; + + /* ERRORS */ +not_negotiated: + { + GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL), + ("decoder not initialized")); + return GST_FLOW_NOT_NEGOTIATED; + } +} + +static GstFlowReturn +gst_speex_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) +{ + GstFlowReturn res; + GstSpeexDec *dec; + + /* no fancy draining */ + if (G_UNLIKELY (!buf)) + return GST_FLOW_OK; + + dec = GST_SPEEX_DEC (bdec); + + /* If we have the streamheader and vorbiscomment from the caps already + * ignore them here */ + if (dec->streamheader && dec->vorbiscomment) { + if (GST_BUFFER_SIZE (dec->streamheader) == GST_BUFFER_SIZE (buf) + && memcmp (GST_BUFFER_DATA (dec->streamheader), GST_BUFFER_DATA (buf), + GST_BUFFER_SIZE (buf)) == 0) { + GST_DEBUG_OBJECT (dec, "found streamheader"); + gst_audio_decoder_finish_frame (bdec, NULL, 1); + res = GST_FLOW_OK; + } else if (GST_BUFFER_SIZE (dec->vorbiscomment) == GST_BUFFER_SIZE (buf) + && memcmp (GST_BUFFER_DATA (dec->vorbiscomment), GST_BUFFER_DATA (buf), + GST_BUFFER_SIZE (buf)) == 0) { + GST_DEBUG_OBJECT (dec, "found vorbiscomments"); + gst_audio_decoder_finish_frame (bdec, NULL, 1); + res = GST_FLOW_OK; + } else { + res = gst_speex_dec_parse_data (dec, buf); + } + } else { + /* Otherwise fall back to packet counting and assume that the + * first two packets are the headers. */ + switch (dec->packetno) { + case 0: + GST_DEBUG_OBJECT (dec, "counted streamheader"); + res = gst_speex_dec_parse_header (dec, buf); + gst_audio_decoder_finish_frame (bdec, NULL, 1); + break; + case 1: + GST_DEBUG_OBJECT (dec, "counted vorbiscomments"); + res = gst_speex_dec_parse_comments (dec, buf); + gst_audio_decoder_finish_frame (bdec, NULL, 1); + break; + default: + { + res = gst_speex_dec_parse_data (dec, buf); + break; + } + } + } + + dec->packetno++; + + return res; +} + +static void +gst_speex_dec_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstSpeexDec *speexdec; + + speexdec = GST_SPEEX_DEC (object); + + switch (prop_id) { + case ARG_ENH: + g_value_set_boolean (value, speexdec->enh); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_speex_dec_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstSpeexDec *speexdec; + + speexdec = GST_SPEEX_DEC (object); + + switch (prop_id) { + case ARG_ENH: + speexdec->enh = g_value_get_boolean (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} diff --git a/ext/speex/gstspeexdec.h b/ext/speex/gstspeexdec.h new file mode 100644 index 0000000..8187af8 --- /dev/null +++ b/ext/speex/gstspeexdec.h @@ -0,0 +1,80 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_SPEEX_DEC_H__ +#define __GST_SPEEX_DEC_H__ + +#include +#include + +#include +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_SPEEX_DEC \ + (gst_speex_dec_get_type()) +#define GST_SPEEX_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_DEC,GstSpeexDec)) +#define GST_SPEEX_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_DEC,GstSpeexDecClass)) +#define GST_IS_SPEEX_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_DEC)) +#define GST_IS_SPEEX_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_DEC)) + +typedef struct _GstSpeexDec GstSpeexDec; +typedef struct _GstSpeexDecClass GstSpeexDecClass; + +struct _GstSpeexDec { + GstAudioDecoder element; + + void *state; + SpeexStereoState *stereo; +#ifdef SPEEX_1_0 + SpeexMode *mode; +#else + const SpeexMode *mode; +#endif + SpeexHeader *header; + SpeexCallback callback; + SpeexBits bits; + + gboolean enh; + + gint frame_size; + GstClockTime frame_duration; + guint64 packetno; + + GstBuffer *streamheader; + GstBuffer *vorbiscomment; +}; + +struct _GstSpeexDecClass { + GstAudioDecoderClass parent_class; +}; + +GType gst_speex_dec_get_type (void); + +G_END_DECLS + +#endif /* __GST_SPEEX_DEC_H__ */ diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c new file mode 100644 index 0000000..b866e5c --- /dev/null +++ b/ext/speex/gstspeexenc.c @@ -0,0 +1,850 @@ +/* GStreamer Speex Encoder + * Copyright (C) <1999> Erik Walthinsen + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-speexenc + * @see_also: speexdec, oggmux + * + * This element encodes audio as a Speex stream. + * Speex is a royalty-free + * audio codec maintained by the Xiph.org + * Foundation. + * + * + * Example pipelines + * |[ + * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg + * ]| Encode an Ogg/Speex file. + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include "gstspeexenc.h" + +GST_DEBUG_CATEGORY_STATIC (speexenc_debug); +#define GST_CAT_DEFAULT speexenc_debug + +static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "rate = (int) [ 6000, 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-speex, " + "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2]") + ); + +#define DEFAULT_QUALITY 8.0 +#define DEFAULT_BITRATE 0 +#define DEFAULT_MODE GST_SPEEX_ENC_MODE_AUTO +#define DEFAULT_VBR FALSE +#define DEFAULT_ABR 0 +#define DEFAULT_VAD FALSE +#define DEFAULT_DTX FALSE +#define DEFAULT_COMPLEXITY 3 +#define DEFAULT_NFRAMES 1 + +enum +{ + PROP_0, + PROP_QUALITY, + PROP_BITRATE, + PROP_MODE, + PROP_VBR, + PROP_ABR, + PROP_VAD, + PROP_DTX, + PROP_COMPLEXITY, + PROP_NFRAMES, + PROP_LAST_MESSAGE +}; + +#define GST_TYPE_SPEEX_ENC_MODE (gst_speex_enc_mode_get_type()) +static GType +gst_speex_enc_mode_get_type (void) +{ + static GType speex_enc_mode_type = 0; + static const GEnumValue speex_enc_modes[] = { + {GST_SPEEX_ENC_MODE_AUTO, "Auto", "auto"}, + {GST_SPEEX_ENC_MODE_UWB, "Ultra Wide Band", "uwb"}, + {GST_SPEEX_ENC_MODE_WB, "Wide Band", "wb"}, + {GST_SPEEX_ENC_MODE_NB, "Narrow Band", "nb"}, + {0, NULL, NULL}, + }; + if (G_UNLIKELY (speex_enc_mode_type == 0)) { + speex_enc_mode_type = g_enum_register_static ("GstSpeexEncMode", + speex_enc_modes); + } + return speex_enc_mode_type; +} + +static void gst_speex_enc_finalize (GObject * object); + +static gboolean gst_speex_enc_setup (GstSpeexEnc * enc); + +static void gst_speex_enc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_speex_enc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); + +static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf); + +static gboolean gst_speex_enc_start (GstAudioEncoder * enc); +static gboolean gst_speex_enc_stop (GstAudioEncoder * enc); +static gboolean gst_speex_enc_set_format (GstAudioEncoder * enc, + GstAudioInfo * info); +static GstFlowReturn gst_speex_enc_handle_frame (GstAudioEncoder * enc, + GstBuffer * in_buf); +static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc, + GstEvent * event); +static GstFlowReturn +gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer); + +static void +gst_speex_enc_setup_interfaces (GType speexenc_type) +{ + static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL }; + + g_type_add_interface_static (speexenc_type, GST_TYPE_TAG_SETTER, + &tag_setter_info); + + GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder"); +} + +GST_BOILERPLATE_FULL (GstSpeexEnc, gst_speex_enc, GstAudioEncoder, + GST_TYPE_AUDIO_ENCODER, gst_speex_enc_setup_interfaces); + +static void +gst_speex_enc_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_static_pad_template (element_class, &src_factory); + gst_element_class_add_static_pad_template (element_class, &sink_factory); + gst_element_class_set_details_simple (element_class, "Speex audio encoder", + "Codec/Encoder/Audio", + "Encodes audio in Speex format", "Wim Taymans "); +} + +static void +gst_speex_enc_class_init (GstSpeexEncClass * klass) +{ + GObjectClass *gobject_class; + GstAudioEncoderClass *base_class; + + gobject_class = (GObjectClass *) klass; + base_class = (GstAudioEncoderClass *) klass; + + gobject_class->set_property = gst_speex_enc_set_property; + gobject_class->get_property = gst_speex_enc_get_property; + + base_class->start = GST_DEBUG_FUNCPTR (gst_speex_enc_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame); + base_class->event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event); + base_class->pre_push = GST_DEBUG_FUNCPTR (gst_speex_enc_pre_push); + + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY, + g_param_spec_float ("quality", "Quality", "Encoding quality", + 0.0, 10.0, DEFAULT_QUALITY, + G_PARAM_CONSTRUCT | 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). (0 = automatic)", + 0, G_MAXINT, DEFAULT_BITRATE, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", "The encoding mode", + GST_TYPE_SPEEX_ENC_MODE, GST_SPEEX_ENC_MODE_AUTO, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VBR, + g_param_spec_boolean ("vbr", "VBR", + "Enable variable bit-rate", DEFAULT_VBR, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ABR, + g_param_spec_int ("abr", "ABR", + "Enable average bit-rate (0 = disabled)", + 0, G_MAXINT, DEFAULT_ABR, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VAD, + g_param_spec_boolean ("vad", "VAD", + "Enable voice activity detection", DEFAULT_VAD, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DTX, + g_param_spec_boolean ("dtx", "DTX", + "Enable discontinuous transmission", DEFAULT_DTX, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLEXITY, + g_param_spec_int ("complexity", "Complexity", + "Set encoding complexity", + 0, G_MAXINT, DEFAULT_COMPLEXITY, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_NFRAMES, + g_param_spec_int ("nframes", "NFrames", + "Number of frames per buffer", + 0, G_MAXINT, DEFAULT_NFRAMES, + G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LAST_MESSAGE, + g_param_spec_string ("last-message", "last-message", + "The last status message", NULL, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + gobject_class->finalize = gst_speex_enc_finalize; +} + +static void +gst_speex_enc_finalize (GObject * object) +{ + GstSpeexEnc *enc; + + enc = GST_SPEEX_ENC (object); + + g_free (enc->last_message); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_speex_enc_init (GstSpeexEnc * enc, GstSpeexEncClass * klass) +{ + GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); + + /* arrange granulepos marking (and required perfect ts) */ + gst_audio_encoder_set_mark_granule (benc, TRUE); + gst_audio_encoder_set_perfect_timestamp (benc, TRUE); +} + +static gboolean +gst_speex_enc_start (GstAudioEncoder * benc) +{ + GstSpeexEnc *enc = GST_SPEEX_ENC (benc); + + GST_DEBUG_OBJECT (enc, "start"); + speex_bits_init (&enc->bits); + enc->tags = gst_tag_list_new (); + enc->header_sent = FALSE; + + return TRUE; +} + +static gboolean +gst_speex_enc_stop (GstAudioEncoder * benc) +{ + GstSpeexEnc *enc = GST_SPEEX_ENC (benc); + + GST_DEBUG_OBJECT (enc, "stop"); + enc->header_sent = FALSE; + if (enc->state) { + speex_encoder_destroy (enc->state); + enc->state = NULL; + } + speex_bits_destroy (&enc->bits); + gst_tag_list_free (enc->tags); + enc->tags = NULL; + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + + gst_tag_setter_reset_tags (GST_TAG_SETTER (enc)); + + return TRUE; +} + +static gint64 +gst_speex_enc_get_latency (GstSpeexEnc * enc) +{ + /* See the Speex manual section "Latency and algorithmic delay" */ + if (enc->rate == 8000) + return 30 * GST_MSECOND; + else + return 34 * GST_MSECOND; +} + +static gboolean +gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) +{ + GstSpeexEnc *enc; + + enc = GST_SPEEX_ENC (benc); + + enc->channels = GST_AUDIO_INFO_CHANNELS (info); + enc->rate = GST_AUDIO_INFO_RATE (info); + + /* handle reconfigure */ + if (enc->state) { + speex_encoder_destroy (enc->state); + enc->state = NULL; + } + + if (!gst_speex_enc_setup (enc)) + return FALSE; + + /* feedback to base class */ + gst_audio_encoder_set_latency (benc, + gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc)); + gst_audio_encoder_set_lookahead (benc, enc->lookahead); + + if (enc->nframes == 0) { + /* as many frames as available input allows */ + gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size); + gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size); + gst_audio_encoder_set_frame_max (benc, 0); + } else { + /* exactly as many frames as configured */ + gst_audio_encoder_set_frame_samples_min (benc, + enc->frame_size * enc->nframes); + gst_audio_encoder_set_frame_samples_max (benc, + enc->frame_size * enc->nframes); + gst_audio_encoder_set_frame_max (benc, 1); + } + + return TRUE; +} + +static GstBuffer * +gst_speex_enc_create_metadata_buffer (GstSpeexEnc * enc) +{ + const GstTagList *user_tags; + GstTagList *merged_tags; + GstBuffer *comments = NULL; + + user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc)); + + GST_DEBUG_OBJECT (enc, "upstream tags = %" GST_PTR_FORMAT, enc->tags); + GST_DEBUG_OBJECT (enc, "user-set tags = %" GST_PTR_FORMAT, user_tags); + + /* gst_tag_list_merge() will handle NULL for either or both lists fine */ + merged_tags = gst_tag_list_merge (user_tags, enc->tags, + gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc))); + + if (merged_tags == NULL) + merged_tags = gst_tag_list_new (); + + GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags); + comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL, + 0, "Encoded with GStreamer Speexenc"); + gst_tag_list_free (merged_tags); + + GST_BUFFER_OFFSET (comments) = 0; + GST_BUFFER_OFFSET_END (comments) = 0; + + return comments; +} + +static void +gst_speex_enc_set_last_msg (GstSpeexEnc * enc, const gchar * msg) +{ + g_free (enc->last_message); + enc->last_message = g_strdup (msg); + GST_WARNING_OBJECT (enc, "%s", msg); + g_object_notify (G_OBJECT (enc), "last-message"); +} + +static gboolean +gst_speex_enc_setup (GstSpeexEnc * enc) +{ + switch (enc->mode) { + case GST_SPEEX_ENC_MODE_UWB: + GST_LOG_OBJECT (enc, "configuring for requested UWB mode"); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB); + break; + case GST_SPEEX_ENC_MODE_WB: + GST_LOG_OBJECT (enc, "configuring for requested WB mode"); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB); + break; + case GST_SPEEX_ENC_MODE_NB: + GST_LOG_OBJECT (enc, "configuring for requested NB mode"); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB); + break; + case GST_SPEEX_ENC_MODE_AUTO: + /* fall through */ + GST_LOG_OBJECT (enc, "finding best mode"); + default: + break; + } + + if (enc->rate > 25000) { + if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) { + GST_LOG_OBJECT (enc, "selected UWB mode for samplerate %d", enc->rate); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB); + } else { + if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_UWB)) { + gst_speex_enc_set_last_msg (enc, + "Warning: suggest to use ultra wide band mode for this rate"); + } + } + } else if (enc->rate > 12500) { + if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) { + GST_LOG_OBJECT (enc, "selected WB mode for samplerate %d", enc->rate); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB); + } else { + if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_WB)) { + gst_speex_enc_set_last_msg (enc, + "Warning: suggest to use wide band mode for this rate"); + } + } + } else { + if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) { + GST_LOG_OBJECT (enc, "selected NB mode for samplerate %d", enc->rate); + enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB); + } else { + if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_NB)) { + gst_speex_enc_set_last_msg (enc, + "Warning: suggest to use narrow band mode for this rate"); + } + } + } + + if (enc->rate != 8000 && enc->rate != 16000 && enc->rate != 32000) { + gst_speex_enc_set_last_msg (enc, + "Warning: speex is optimized for 8, 16 and 32 KHz"); + } + + speex_init_header (&enc->header, enc->rate, 1, enc->speex_mode); + enc->header.frames_per_packet = enc->nframes; + enc->header.vbr = enc->vbr; + enc->header.nb_channels = enc->channels; + + /*Initialize Speex encoder */ + enc->state = speex_encoder_init (enc->speex_mode); + + speex_encoder_ctl (enc->state, SPEEX_GET_FRAME_SIZE, &enc->frame_size); + speex_encoder_ctl (enc->state, SPEEX_SET_COMPLEXITY, &enc->complexity); + speex_encoder_ctl (enc->state, SPEEX_SET_SAMPLING_RATE, &enc->rate); + + if (enc->vbr) + speex_encoder_ctl (enc->state, SPEEX_SET_VBR_QUALITY, &enc->quality); + else { + gint tmp = floor (enc->quality); + + speex_encoder_ctl (enc->state, SPEEX_SET_QUALITY, &tmp); + } + if (enc->bitrate) { + if (enc->quality >= 0.0 && enc->vbr) { + gst_speex_enc_set_last_msg (enc, + "Warning: bitrate option is overriding quality"); + } + speex_encoder_ctl (enc->state, SPEEX_SET_BITRATE, &enc->bitrate); + } + if (enc->vbr) { + gint tmp = 1; + + speex_encoder_ctl (enc->state, SPEEX_SET_VBR, &tmp); + } else if (enc->vad) { + gint tmp = 1; + + speex_encoder_ctl (enc->state, SPEEX_SET_VAD, &tmp); + } + + if (enc->dtx) { + gint tmp = 1; + + speex_encoder_ctl (enc->state, SPEEX_SET_DTX, &tmp); + } + + if (enc->dtx && !(enc->vbr || enc->abr || enc->vad)) { + gst_speex_enc_set_last_msg (enc, + "Warning: dtx is useless without vad, vbr or abr"); + } else if ((enc->vbr || enc->abr) && (enc->vad)) { + gst_speex_enc_set_last_msg (enc, + "Warning: vad is already implied by vbr or abr"); + } + + if (enc->abr) { + speex_encoder_ctl (enc->state, SPEEX_SET_ABR, &enc->abr); + } + + speex_encoder_ctl (enc->state, SPEEX_GET_LOOKAHEAD, &enc->lookahead); + + GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size, + enc->lookahead); + + return TRUE; +} + +/* push out the buffer */ +static GstFlowReturn +gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer) +{ + guint size; + + size = GST_BUFFER_SIZE (buffer); + GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size); + + gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc))); + return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer); +} + +static gboolean +gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) +{ + GstSpeexEnc *enc; + + enc = GST_SPEEX_ENC (benc); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_TAG: + { + if (enc->tags) { + GstTagList *list; + + gst_event_parse_tag (event, &list); + gst_tag_list_insert (enc->tags, list, + gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc))); + } else { + g_assert_not_reached (); + } + break; + } + default: + break; + } + + /* we only peeked, let base class handle it */ + return FALSE; +} + +static GstFlowReturn +gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf) +{ + gint frame_size = enc->frame_size; + gint bytes = frame_size * 2 * enc->channels, samples, size; + gint outsize, written, dtx_ret = 0; + guint8 *data, *data0 = NULL; + GstBuffer *outbuf; + GstFlowReturn ret = GST_FLOW_OK; + + if (G_LIKELY (buf)) { + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + + if (G_UNLIKELY (size % bytes)) { + GST_DEBUG_OBJECT (enc, "draining; adding silence samples"); + size = ((size / bytes) + 1) * bytes; + data0 = data = g_malloc0 (size); + memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + } + } else { + GST_DEBUG_OBJECT (enc, "nothing to drain"); + goto done; + } + + samples = size / (2 * enc->channels); + speex_bits_reset (&enc->bits); + + /* FIXME what about dropped samples if DTS enabled ?? */ + + while (size) { + GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes); + + if (enc->channels == 2) { + speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits); + } + dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits); + + data += bytes; + size -= bytes; + } + + speex_bits_insert_terminator (&enc->bits); + outsize = speex_bits_nbytes (&enc->bits); + + ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), + GST_BUFFER_OFFSET_NONE, outsize, + GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf); + + if ((GST_FLOW_OK != ret)) + goto done; + + written = speex_bits_write (&enc->bits, + (gchar *) GST_BUFFER_DATA (outbuf), outsize); + + if (G_UNLIKELY (written < outsize)) { + GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize); + GST_BUFFER_SIZE (outbuf) = written; + } else if (G_UNLIKELY (written > outsize)) { + GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize); + } + + if (!dtx_ret) + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); + + ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), + outbuf, samples); + +done: + g_free (data0); + return ret; +} + +/* + * (really really) FIXME: move into core (dixit tpm) + */ +/** + * _gst_caps_set_buffer_array: + * @caps: a #GstCaps + * @field: field in caps to set + * @buf: header buffers + * + * Adds given buffers to an array of buffers set as the given @field + * on the given @caps. List of buffer arguments must be NULL-terminated. + * + * Returns: input caps with a streamheader field added, or NULL if some error + */ +static GstCaps * +_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field, + GstBuffer * buf, ...) +{ + GstStructure *structure = NULL; + va_list va; + GValue array = { 0 }; + GValue value = { 0 }; + + g_return_val_if_fail (caps != NULL, NULL); + g_return_val_if_fail (gst_caps_is_fixed (caps), NULL); + g_return_val_if_fail (field != NULL, NULL); + + caps = gst_caps_make_writable (caps); + structure = gst_caps_get_structure (caps, 0); + + g_value_init (&array, GST_TYPE_ARRAY); + + va_start (va, buf); + /* put buffers in a fixed list */ + while (buf) { + g_assert (gst_buffer_is_metadata_writable (buf)); + + /* mark buffer */ + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); + + g_value_init (&value, GST_TYPE_BUFFER); + buf = gst_buffer_copy (buf); + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); + gst_value_set_buffer (&value, buf); + gst_buffer_unref (buf); + gst_value_array_append_value (&array, &value); + g_value_unset (&value); + + buf = va_arg (va, GstBuffer *); + } + + gst_structure_set_value (structure, field, &array); + g_value_unset (&array); + + return caps; +} + +static GstFlowReturn +gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) +{ + GstSpeexEnc *enc; + GstFlowReturn ret = GST_FLOW_OK; + + enc = GST_SPEEX_ENC (benc); + + if (!enc->header_sent) { + /* Speex 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 libspeex + one at a time; libspeex handles the additional Ogg bitstream + constraints */ + GstBuffer *buf1, *buf2; + GstCaps *caps; + guchar *data; + gint data_len; + + /* create header buffer */ + data = (guint8 *) speex_header_to_packet (&enc->header, &data_len); + buf1 = gst_buffer_new (); + GST_BUFFER_DATA (buf1) = GST_BUFFER_MALLOCDATA (buf1) = data; + GST_BUFFER_SIZE (buf1) = data_len; + GST_BUFFER_OFFSET_END (buf1) = 0; + GST_BUFFER_OFFSET (buf1) = 0; + + /* create comment buffer */ + buf2 = gst_speex_enc_create_metadata_buffer (enc); + + /* mark and put on caps */ + caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, enc->rate, + "channels", G_TYPE_INT, enc->channels, NULL); + caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL); + + /* negotiate with these caps */ + GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps); + + gst_buffer_set_caps (buf1, caps); + gst_buffer_set_caps (buf2, caps); + gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps); + gst_caps_unref (caps); + + /* push out buffers */ + /* store buffers for later pre_push sending */ + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + GST_DEBUG_OBJECT (enc, "storing header buffers"); + enc->headers = g_slist_prepend (enc->headers, buf2); + enc->headers = g_slist_prepend (enc->headers, buf1); + + enc->header_sent = TRUE; + } + + GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf, + buf ? GST_BUFFER_SIZE (buf) : 0); + + ret = gst_speex_enc_encode (enc, buf); + + return ret; +} + +static GstFlowReturn +gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer) +{ + GstSpeexEnc *enc; + GstFlowReturn ret = GST_FLOW_OK; + + enc = GST_SPEEX_ENC (benc); + + /* FIXME 0.11 ? get rid of this special ogg stuff and have it + * put and use 'codec data' in caps like anything else, + * with all the usual out-of-band advantage etc */ + if (G_UNLIKELY (enc->headers)) { + GSList *header = enc->headers; + + /* try to push all of these, if we lose one, might as well lose all */ + while (header) { + if (ret == GST_FLOW_OK) + ret = gst_speex_enc_push_buffer (enc, header->data); + else + gst_speex_enc_push_buffer (enc, header->data); + header = g_slist_next (header); + } + + g_slist_free (enc->headers); + enc->headers = NULL; + } + + return ret; +} + +static void +gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value, + GParamSpec * pspec) +{ + GstSpeexEnc *enc; + + enc = GST_SPEEX_ENC (object); + + switch (prop_id) { + case PROP_QUALITY: + g_value_set_float (value, enc->quality); + break; + case PROP_BITRATE: + g_value_set_int (value, enc->bitrate); + break; + case PROP_MODE: + g_value_set_enum (value, enc->mode); + break; + case PROP_VBR: + g_value_set_boolean (value, enc->vbr); + break; + case PROP_ABR: + g_value_set_int (value, enc->abr); + break; + case PROP_VAD: + g_value_set_boolean (value, enc->vad); + break; + case PROP_DTX: + g_value_set_boolean (value, enc->dtx); + break; + case PROP_COMPLEXITY: + g_value_set_int (value, enc->complexity); + break; + case PROP_NFRAMES: + g_value_set_int (value, enc->nframes); + break; + case PROP_LAST_MESSAGE: + g_value_set_string (value, enc->last_message); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_speex_enc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstSpeexEnc *enc; + + enc = GST_SPEEX_ENC (object); + + switch (prop_id) { + case PROP_QUALITY: + enc->quality = g_value_get_float (value); + break; + case PROP_BITRATE: + enc->bitrate = g_value_get_int (value); + break; + case PROP_MODE: + enc->mode = g_value_get_enum (value); + break; + case PROP_VBR: + enc->vbr = g_value_get_boolean (value); + break; + case PROP_ABR: + enc->abr = g_value_get_int (value); + break; + case PROP_VAD: + enc->vad = g_value_get_boolean (value); + break; + case PROP_DTX: + enc->dtx = g_value_get_boolean (value); + break; + case PROP_COMPLEXITY: + enc->complexity = g_value_get_int (value); + break; + case PROP_NFRAMES: + enc->nframes = g_value_get_int (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} diff --git a/ext/speex/gstspeexenc.h b/ext/speex/gstspeexenc.h new file mode 100644 index 0000000..5cf5cd8 --- /dev/null +++ b/ext/speex/gstspeexenc.h @@ -0,0 +1,102 @@ +/* GStreamer Speex Encoder + * Copyright (C) <1999> Erik Walthinsen + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_SPEEX_ENC_H__ +#define __GST_SPEEX_ENC_H__ + + +#include +#include + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_SPEEX_ENC \ + (gst_speex_enc_get_type()) +#define GST_SPEEX_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_ENC,GstSpeexEnc)) +#define GST_SPEEX_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_ENC,GstSpeexEncClass)) +#define GST_IS_SPEEX_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_ENC)) +#define GST_IS_SPEEX_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_ENC)) + +typedef enum +{ + GST_SPEEX_ENC_MODE_AUTO, + GST_SPEEX_ENC_MODE_UWB, + GST_SPEEX_ENC_MODE_WB, + GST_SPEEX_ENC_MODE_NB +} GstSpeexMode; + +typedef struct _GstSpeexEnc GstSpeexEnc; +typedef struct _GstSpeexEncClass GstSpeexEncClass; + +struct _GstSpeexEnc { + GstAudioEncoder element; + + SpeexBits bits; + SpeexHeader header; +#ifdef SPEEX_1_0 + SpeexMode *speex_mode; +#else + const SpeexMode *speex_mode; +#endif + void *state; + + /* properties */ + GstSpeexMode mode; + gfloat quality; + gint bitrate; + gboolean vbr; + gint abr; + gboolean vad; + gboolean dtx; + gint complexity; + gint nframes; + gchar *last_message; + + gint channels; + gint rate; + + gboolean header_sent; + GSList *headers; + + GstTagList *tags; + + gint frame_size; + gint lookahead; + + guint8 *comments; + gint comment_len; +}; + +struct _GstSpeexEncClass { + GstAudioEncoderClass parent_class; +}; + +GType gst_speex_enc_get_type (void); + +G_END_DECLS + +#endif /* __GST_SPEEXENC_H__ */ -- cgit v1.2.3