/* GStreamer * Copyright (C) 2004 Wim Taymans <wim@fluendo.com> * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net> * * 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. * <ulink url="http://www.speex.org/">Speex</ulink> is a royalty-free * audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org * Foundation</ulink>. * * <refsect2> * <title>Example pipelines</title> * |[ * 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. * </refsect2> * * Last reviewed on 2006-04-05 (0.10.2) */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "gstspeexdec.h" #include <stdlib.h> #include <string.h> #include <gst/tag/tag.h> #include <gst/audio/audio.h> GST_DEBUG_CATEGORY_STATIC (speexdec_debug); #define GST_CAT_DEFAULT speexdec_debug #define DEFAULT_ENH TRUE enum { ARG_0, ARG_ENH }; #define FORMAT_STR GST_AUDIO_NE(S16) static GstStaticPadTemplate speex_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " FORMAT_STR ", " "layout = (string) interleaved, " "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]") ); static GstStaticPadTemplate speex_dec_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-speex") ); #define gst_speex_dec_parent_class parent_class G_DEFINE_TYPE (GstSpeexDec, gst_speex_dec, 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_class_init (GstSpeexDecClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstAudioDecoderClass *base_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) 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_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&speex_dec_src_factory)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&speex_dec_sink_factory)); gst_element_class_set_details_simple (gstelement_class, "Speex audio decoder", "Codec/Decoder/Audio", "decode speex streams to audio", "Wim Taymans <wim@fluendo.com>"); 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) { 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) { GstMapInfo map; GstAudioInfo info; static const GstAudioChannelPosition chan_pos[2][2] = { {GST_AUDIO_CHANNEL_POSITION_MONO}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT} }; /* get the header */ gst_buffer_map (buf, &map, GST_MAP_READ); dec->header = speex_packet_to_header ((gchar *) map.data, map.size); gst_buffer_unmap (buf, &map); 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 */ gst_audio_info_init (&info); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, dec->header->rate, dec->header->nb_channels, chan_pos[dec->header->nb_channels - 1]); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info)) goto nego_failed; 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")); 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_empty (); } 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_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (dec), gst_event_new_tag (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; SpeexBits *bits; GstMapInfo map; if (!dec->frame_duration) goto not_negotiated; if (G_LIKELY (gst_buffer_get_size (buf))) { /* send data to the bitstream */ gst_buffer_map (buf, &map, GST_MAP_READ); speex_bits_read_from (&dec->bits, (gchar *) map.data, map.size); gst_buffer_unmap (buf, &map); fpp = dec->header->frames_per_packet; bits = &dec->bits; GST_DEBUG_OBJECT (dec, "received buffer of size %" G_GSIZE_FORMAT ", fpp %d, %d bits", map.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; gint ret; GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp, bits ? speex_bits_remaining (bits) : -1); #if 0 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; } #endif /* FIXME, we can use a bufferpool because we have fixed size buffers. We * could also use an allocator */ outbuf = gst_buffer_new_allocate (NULL, dec->frame_size * dec->header->nb_channels * 2, 0); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); ret = speex_decode_int (dec->state, bits, (spx_int16_t *) map.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 ((spx_int16_t *) map.data, dec->frame_size, dec->stereo); gst_buffer_unmap (outbuf, &map); 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 gboolean memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2) { GstMapInfo map; gsize size1, size2; gboolean res; size1 = gst_buffer_get_size (buf1); size2 = gst_buffer_get_size (buf2); if (size1 != size2) return FALSE; gst_buffer_map (buf1, &map, GST_MAP_READ); res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0; gst_buffer_unmap (buf1, &map); return res; } 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 (memcmp_buffers (dec->streamheader, buf)) { GST_DEBUG_OBJECT (dec, "found streamheader"); gst_audio_decoder_finish_frame (bdec, NULL, 1); res = GST_FLOW_OK; } else if (memcmp_buffers (dec->vorbiscomment, buf)) { 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; } }