/* GStreamer FAAD (Free AAC Decoder) plugin * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-faad * @seealso: faac * * faad decodes AAC (MPEG-4 part 3) stream. * * <refsect2> * <title>Example launch lines</title> * |[ * gst-launch filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink * ]| Play aac from mp4 file. * |[ * gst-launch filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink * ]| Play standalone aac bitstream. * </refsect2> */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include <gst/audio/audio.h> /* These are the correct types for these functions, as defined in the source, * with types changed to match glib types, since those are defined for us. * However, upstream FAAD is distributed with a broken header file that defined * these wrongly (in a way which was broken on 64 bit systems). * * Upstream CVS still has the bug, but has also renamed all the public symbols * for Better Corporate Branding (or whatever), so we need to take that * (FAAD_IS_NEAAC) into account as well. * * We must call them using these definitions. Most distributions now have the * corrected header file (they distribute a patch along with the source), * but not all, hence this Truly Evil Hack. * * Note: The prototypes don't need to be defined conditionaly, as the cpp will * do that for us. */ #if FAAD2_MINOR_VERSION < 7 #ifdef FAAD_IS_NEAAC #define NeAACDecInit NeAACDecInit_no_definition #define NeAACDecInit2 NeAACDecInit2_no_definition #else #define faacDecInit faacDecInit_no_definition #define faacDecInit2 faacDecInit2_no_definition #endif #endif /* FAAD2_MINOR_VERSION < 7 */ #include "gstfaad.h" #if FAAD2_MINOR_VERSION < 7 #ifdef FAAD_IS_NEAAC #undef NeAACDecInit #undef NeAACDecInit2 #else #undef faacDecInit #undef faacDecInit2 #endif extern long faacDecInit (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *); extern gint8 faacDecInit2 (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *); #endif /* FAAD2_MINOR_VERSION < 7 */ GST_DEBUG_CATEGORY_STATIC (faad_debug); #define GST_CAT_DEFAULT faad_debug static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 2; " "audio/mpeg, mpegversion = (int) 4, stream-format = (string) { raw, adts }") ); #define STATIC_RAW_CAPS(format) \ "audio/x-raw, " \ "format = (string) "GST_AUDIO_NE(format)", " \ "layout = (string) interleaved, " \ "rate = (int) [ 8000, 96000 ], " \ "channels = (int) [ 1, 8 ]" /* * All except 16-bit integer are disabled until someone fixes FAAD. * FAAD allocates approximately 8*1024*2 bytes bytes, which is enough * for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp * audio, but not for any other. You'll get random segfaults, crashes * and even valgrind goes crazy. */ #define STATIC_CAPS \ STATIC_RAW_CAPS (S16) #if 0 #define NOTUSED "; " \ STATIC_RAW_CAPS (S24) \ "; " \ STATIC_RAW_CAPS (S32) \ "; " \ STATIC_RAW_CAPS (F32) \ "; " \ STATIC_RAW_CAPS (F64) #endif static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (STATIC_CAPS) ); static void gst_faad_reset (GstFaad * faad); static gboolean gst_faad_start (GstAudioDecoder * dec); static gboolean gst_faad_stop (GstAudioDecoder * dec); static gboolean gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps); static gboolean gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length); static GstFlowReturn gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer); static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard); static gboolean gst_faad_open_decoder (GstFaad * faad); static void gst_faad_close_decoder (GstFaad * faad); #define gst_faad_parent_class parent_class G_DEFINE_TYPE (GstFaad, gst_faad, GST_TYPE_AUDIO_DECODER); static void gst_faad_class_init (GstFaadClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); gst_element_class_set_static_metadata (element_class, "AAC audio decoder", "Codec/Decoder/Audio", "Free MPEG-2/4 AAC decoder", "Ronald Bultje <rbultje@ronald.bitfreak.net>"); base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format); base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame); base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush); GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding"); } static void gst_faad_init (GstFaad * faad) { gst_faad_reset (faad); } static void gst_faad_reset_stream_state (GstFaad * faad) { if (faad->handle) faacDecPostSeekReset (faad->handle, 0); } static void gst_faad_reset (GstFaad * faad) { faad->samplerate = -1; faad->channels = -1; faad->init = FALSE; faad->packetised = FALSE; g_free (faad->channel_positions); faad->channel_positions = NULL; faad->last_header = 0; gst_faad_reset_stream_state (faad); } static gboolean gst_faad_start (GstAudioDecoder * dec) { GstFaad *faad = GST_FAAD (dec); GST_DEBUG_OBJECT (dec, "start"); gst_faad_reset (faad); /* call upon legacy upstream byte support (e.g. seeking) */ gst_audio_decoder_set_estimate_rate (dec, TRUE); /* never mind a few errors */ gst_audio_decoder_set_max_errors (dec, 10); return TRUE; } static gboolean gst_faad_stop (GstAudioDecoder * dec) { GstFaad *faad = GST_FAAD (dec); GST_DEBUG_OBJECT (dec, "stop"); gst_faad_reset (faad); gst_faad_close_decoder (faad); return TRUE; } static gint aac_rate_idx (gint rate) { if (92017 <= rate) return 0; else if (75132 <= rate) return 1; else if (55426 <= rate) return 2; else if (46009 <= rate) return 3; else if (37566 <= rate) return 4; else if (27713 <= rate) return 5; else if (23004 <= rate) return 6; else if (18783 <= rate) return 7; else if (13856 <= rate) return 8; else if (11502 <= rate) return 9; else if (9391 <= rate) return 10; else return 11; } static gboolean gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps) { GstFaad *faad = GST_FAAD (dec); GstStructure *str = gst_caps_get_structure (caps, 0); GstBuffer *buf; const GValue *value; GstMapInfo map; guint8 *cdata; gsize csize; /* clean up current decoder, rather than trying to reconfigure */ gst_faad_close_decoder (faad); /* Assume raw stream */ faad->packetised = FALSE; if ((value = gst_structure_get_value (str, "codec_data"))) { #if FAAD2_MINOR_VERSION >= 7 unsigned long samplerate; #else guint32 samplerate; #endif guint8 channels; /* We have codec data, means packetised stream */ faad->packetised = TRUE; buf = gst_value_get_buffer (value); g_return_val_if_fail (buf != NULL, FALSE); gst_buffer_map (buf, &map, GST_MAP_READ); cdata = map.data; csize = map.size; if (csize < 2) goto wrong_length; GST_DEBUG_OBJECT (faad, "codec_data: object_type=%d, sample_rate=%d, channels=%d", ((cdata[0] & 0xf8) >> 3), (((cdata[0] & 0x07) << 1) | ((cdata[1] & 0x80) >> 7)), ((cdata[1] & 0x78) >> 3)); if (!gst_faad_open_decoder (faad)) goto open_failed; /* someone forgot that char can be unsigned when writing the API */ if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate, &channels) < 0) goto init_failed; if (channels != ((cdata[1] & 0x78) >> 3)) { /* https://bugs.launchpad.net/ubuntu/+source/faad2/+bug/290259 */ GST_WARNING_OBJECT (faad, "buggy faad version, wrong nr of channels %d instead of %d", channels, ((cdata[1] & 0x78) >> 3)); } GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels, (guint32) samplerate); /* not updating these here, so they are updated in the * chain function, and new caps are created etc. */ faad->samplerate = 0; faad->channels = 0; faad->init = TRUE; gst_buffer_unmap (buf, &map); } else if ((value = gst_structure_get_value (str, "framed")) && g_value_get_boolean (value) == TRUE) { faad->packetised = TRUE; faad->init = FALSE; GST_DEBUG_OBJECT (faad, "we have packetized audio"); } else { faad->init = FALSE; } faad->fake_codec_data[0] = 0; faad->fake_codec_data[1] = 0; if (faad->packetised && !faad->init) { gint rate, channels; if (gst_structure_get_int (str, "rate", &rate) && gst_structure_get_int (str, "channels", &channels)) { gint rate_idx, profile; profile = 3; /* 0=MAIN, 1=LC, 2=SSR, 3=LTP */ rate_idx = aac_rate_idx (rate); faad->fake_codec_data[0] = ((profile + 1) << 3) | ((rate_idx & 0xE) >> 1); faad->fake_codec_data[1] = ((rate_idx & 0x1) << 7) | (channels << 3); GST_LOG_OBJECT (faad, "created fake codec data (%u,%u): 0x%x 0x%x", rate, channels, (int) faad->fake_codec_data[0], (int) faad->fake_codec_data[1]); } } return TRUE; /* ERRORS */ wrong_length: { GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long"); gst_object_unref (faad); gst_buffer_unmap (buf, &map); return FALSE; } open_failed: { GST_DEBUG_OBJECT (faad, "failed to create decoder"); gst_object_unref (faad); gst_buffer_unmap (buf, &map); return FALSE; } init_failed: { GST_DEBUG_OBJECT (faad, "faacDecInit2() failed"); gst_object_unref (faad); gst_buffer_unmap (buf, &map); return FALSE; } } static gboolean gst_faad_chanpos_to_gst (GstFaad * faad, guchar * fpos, GstAudioChannelPosition * pos, guint num) { guint n; gboolean unknown_channel = FALSE; /* special handling for the common cases for mono and stereo */ if (num == 1 && fpos[0] == FRONT_CHANNEL_CENTER) { GST_DEBUG_OBJECT (faad, "mono common case; won't set channel positions"); pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; return TRUE; } else if (num == 2 && fpos[0] == FRONT_CHANNEL_LEFT && fpos[1] == FRONT_CHANNEL_RIGHT) { GST_DEBUG_OBJECT (faad, "stereo common case; won't set channel positions"); pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; return TRUE; } for (n = 0; n < num; n++) { GST_DEBUG_OBJECT (faad, "faad channel %d as %d", n, fpos[n]); switch (fpos[n]) { case FRONT_CHANNEL_LEFT: pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; break; case FRONT_CHANNEL_RIGHT: pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; break; case FRONT_CHANNEL_CENTER: /* argh, mono = center */ if (num == 1) pos[n] = GST_AUDIO_CHANNEL_POSITION_MONO; else pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; break; case SIDE_CHANNEL_LEFT: pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT; break; case SIDE_CHANNEL_RIGHT: pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT; break; case BACK_CHANNEL_LEFT: pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; break; case BACK_CHANNEL_RIGHT: pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; break; case BACK_CHANNEL_CENTER: pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; break; case LFE_CHANNEL: pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE1; break; default: GST_DEBUG_OBJECT (faad, "unknown channel %d at %d", fpos[n], n); unknown_channel = TRUE; break; } } if (unknown_channel) { switch (num) { case 1:{ GST_DEBUG_OBJECT (faad, "FAAD reports unknown 1 channel mapping. Forcing to mono"); pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; break; } case 2:{ GST_DEBUG_OBJECT (faad, "FAAD reports unknown 2 channel mapping. Forcing to stereo"); pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; break; } default:{ GST_WARNING_OBJECT (faad, "Unsupported FAAD channel position 0x%x encountered", fpos[n]); return FALSE; break; } } } return TRUE; } static gboolean gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info) { gboolean ret; gboolean fmt_change = FALSE; GstAudioInfo ainfo; gint i; /* see if we need to renegotiate */ if (info->samplerate != faad->samplerate || info->channels != faad->channels || !faad->channel_positions) { fmt_change = TRUE; } else { for (i = 0; i < info->channels; i++) { if (info->channel_position[i] != faad->channel_positions[i]) { fmt_change = TRUE; break; } } } if (G_LIKELY (gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (faad)) && !fmt_change)) return TRUE; /* store new negotiation information */ faad->samplerate = info->samplerate; faad->channels = info->channels; g_free (faad->channel_positions); faad->channel_positions = g_memdup (info->channel_position, faad->channels); /* FIXME: Use the GstAudioInfo of GstAudioDecoder for all of this */ gst_audio_info_init (&ainfo); gst_audio_info_set_format (&ainfo, GST_AUDIO_FORMAT_S16, faad->samplerate, faad->channels, NULL); faad->bps = 16 / 8; if (!gst_faad_chanpos_to_gst (faad, faad->channel_positions, faad->aac_positions, faad->channels)) { GST_DEBUG_OBJECT (faad, "Could not map channel positions"); return FALSE; } memcpy (ainfo.position, faad->aac_positions, faad->channels * sizeof (GstAudioChannelPosition)); gst_audio_channel_positions_to_valid_order (ainfo.position, faad->channels); memcpy (faad->gst_positions, ainfo.position, faad->channels * sizeof (GstAudioChannelPosition)); /* Unset UNPOSITIONED flag */ if (ainfo.position[0] != GST_AUDIO_CHANNEL_POSITION_NONE) ainfo.flags &= ~GST_AUDIO_FLAG_UNPOSITIONED; /* get the remap table */ memset (faad->reorder_map, 0, sizeof (faad->reorder_map)); faad->need_reorder = FALSE; if (gst_audio_get_channel_reorder_map (faad->channels, faad->aac_positions, faad->gst_positions, faad->reorder_map)) { for (i = 0; i < faad->channels; i++) { GST_DEBUG_OBJECT (faad, "remap %d -> %d", i, faad->reorder_map[i]); if (faad->reorder_map[i] != i) { faad->need_reorder = TRUE; } } } ret = gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (faad), &ainfo); return ret; } /* * Find syncpoint in ADTS/ADIF stream. Doesn't work for raw, * packetized streams. Be careful when calling. * Returns FALSE on no-sync, fills offset/length if one/two * syncpoints are found, only returns TRUE when it finds two * subsequent syncpoints (similar to mp3 typefinding in * gst/typefind/) for ADTS because 12 bits isn't very reliable. */ static gboolean gst_faad_sync (GstFaad * faad, const guint8 * data, guint size, gboolean next, gint * off, gint * length) { guint n = 0; gint snc; gboolean ret = FALSE; guint len = 0; GST_LOG_OBJECT (faad, "Finding syncpoint"); /* check for too small a buffer */ if (size < 3) goto exit; for (n = 0; n < size - 3; n++) { snc = GST_READ_UINT16_BE (&data[n]); if ((snc & 0xfff6) == 0xfff0) { /* we have an ADTS syncpoint. Parse length and find * next syncpoint. */ GST_LOG_OBJECT (faad, "Found one ADTS syncpoint at offset 0x%x, tracing next...", n); if (size - n < 5) { GST_LOG_OBJECT (faad, "Not enough data to parse ADTS header"); break; } len = ((data[n + 3] & 0x03) << 11) | (data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5); if (n + len + 2 >= size) { GST_LOG_OBJECT (faad, "Frame size %d, next frame is not within reach", len); if (next) { break; } else if (n + len <= size) { GST_LOG_OBJECT (faad, "but have complete frame and no next frame; " "accept ADTS syncpoint at offset 0x%x (framelen %u)", n, len); ret = TRUE; break; } } snc = GST_READ_UINT16_BE (&data[n + len]); if ((snc & 0xfff6) == 0xfff0) { GST_LOG_OBJECT (faad, "Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len); ret = TRUE; break; } GST_LOG_OBJECT (faad, "No next frame found... (should be at 0x%x)", n + len); } else if (!memcmp (&data[n], "ADIF", 4)) { /* we have an ADIF syncpoint. 4 bytes is enough. */ GST_LOG_OBJECT (faad, "Found ADIF syncpoint at offset 0x%x", n); ret = TRUE; break; } } exit: *off = n; if (ret) { *length = len; } else { GST_LOG_OBJECT (faad, "Found no syncpoint"); } return ret; } static gboolean looks_like_valid_header (guint8 * input_data, guint input_size) { if (input_size < 4) return FALSE; if (input_data[0] == 'A' && input_data[1] == 'D' && input_data[2] == 'I' && input_data[3] == 'F') /* ADIF type header */ return TRUE; if (input_data[0] == 0xff && (input_data[1] >> 4) == 0xf) /* ADTS type header */ return TRUE; return FALSE; } static GstFlowReturn gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length) { GstFaad *faad; const guint8 *data; guint size; gboolean sync, eos; faad = GST_FAAD (dec); size = gst_adapter_available (adapter); g_return_val_if_fail (size > 0, GST_FLOW_ERROR); gst_audio_decoder_get_parse_state (dec, &sync, &eos); if (faad->packetised) { *offset = 0; *length = size; return GST_FLOW_OK; } else { gboolean ret; data = gst_adapter_map (adapter, size); ret = gst_faad_sync (faad, data, size, !eos, offset, length); gst_adapter_unmap (adapter); return (ret ? GST_FLOW_OK : GST_FLOW_EOS); } } static GstFlowReturn gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) { GstFaad *faad; GstFlowReturn ret = GST_FLOW_OK; GstMapInfo map; gsize input_size; guchar *input_data; GstBuffer *outbuf; faacDecFrameInfo info; void *out; faad = GST_FAAD (dec); /* no fancy draining */ if (G_UNLIKELY (!buffer)) return GST_FLOW_OK; gst_buffer_map (buffer, &map, GST_MAP_READ); input_data = map.data; input_size = map.size; init: /* init if not already done during capsnego */ if (!faad->init) { #if FAAD2_MINOR_VERSION >= 7 unsigned long rate; #else guint32 rate; #endif guint8 ch; GST_DEBUG_OBJECT (faad, "initialising ..."); if (!gst_faad_open_decoder (faad)) goto open_failed; /* We check if the first data looks like it might plausibly contain * appropriate initialisation info... if not, we use our fake_codec_data */ if (looks_like_valid_header (input_data, input_size) || !faad->packetised) { if (faacDecInit (faad->handle, input_data, input_size, &rate, &ch) < 0) goto init_failed; GST_DEBUG_OBJECT (faad, "faacDecInit() ok: rate=%u,channels=%u", (guint32) rate, ch); } else { if ((gint8) faacDecInit2 (faad->handle, faad->fake_codec_data, 2, &rate, &ch) < 0) { goto init2_failed; } GST_DEBUG_OBJECT (faad, "faacDecInit2() ok: rate=%u,channels=%u", (guint32) rate, ch); } faad->init = TRUE; /* make sure we create new caps below */ faad->samplerate = 0; faad->channels = 0; } /* decode cycle */ info.error = 0; do { GstMapInfo omap; if (!faad->packetised) { /* faad only really parses ADTS header at Init time, not when decoding, * so monitor for changes and kick faad when needed */ if (GST_READ_UINT32_BE (input_data) >> 4 != faad->last_header >> 4) { GST_DEBUG_OBJECT (faad, "ADTS header changed, forcing Init"); faad->last_header = GST_READ_UINT32_BE (input_data); /* kick hard */ gst_faad_close_decoder (faad); faad->init = FALSE; goto init; } } out = faacDecDecode (faad->handle, &info, input_data, input_size); gst_buffer_unmap (buffer, &map); buffer = NULL; if (info.error > 0) { /* give up on frame and bail out */ gst_audio_decoder_finish_frame (dec, NULL, 1); goto decode_failed; } GST_LOG_OBJECT (faad, "%d bytes consumed, %d samples decoded", (guint) info.bytesconsumed, (guint) info.samples); if (out && info.samples > 0) { guint channels, samples; if (!gst_faad_update_caps (faad, &info)) goto negotiation_failed; /* C's lovely propensity for int overflow.. */ if (info.samples > G_MAXUINT / faad->bps) goto sample_overflow; channels = faad->channels; /* note: info.samples is total samples, not per channel */ samples = info.samples / channels; /* FIXME, add bufferpool and allocator support to the base class */ outbuf = gst_buffer_new_allocate (NULL, info.samples * faad->bps, NULL); gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE); if (faad->need_reorder) { gint16 *dest, *src, i, j; dest = (gint16 *) omap.data; src = (gint16 *) out; for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { dest[faad->reorder_map[j]] = *src++; } dest += channels; } } else { memcpy (omap.data, out, omap.size); } gst_buffer_unmap (outbuf, &omap); ret = gst_audio_decoder_finish_frame (dec, outbuf, 1); } } while (FALSE); out: if (buffer) gst_buffer_unmap (buffer, &map); return ret; /* ERRORS */ open_failed: { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), ("Failed to open decoder")); ret = GST_FLOW_ERROR; goto out; } init_failed: { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), ("Failed to init decoder from stream")); ret = GST_FLOW_ERROR; goto out; } init2_failed: { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), ("%s() failed", (faad->handle) ? "faacDecInit2" : "faacDecOpen")); ret = GST_FLOW_ERROR; goto out; } decode_failed: { GST_AUDIO_DECODER_ERROR (faad, 1, STREAM, DECODE, (NULL), ("decoding error: %s", faacDecGetErrorMessage (info.error)), ret); goto out; } negotiation_failed: { GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), ("Setting caps on source pad failed")); ret = GST_FLOW_ERROR; goto out; } sample_overflow: { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), ("Output buffer too large")); ret = GST_FLOW_ERROR; goto out; } } static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard) { gst_faad_reset_stream_state (GST_FAAD (dec)); } static gboolean gst_faad_open_decoder (GstFaad * faad) { faacDecConfiguration *conf; faad->handle = faacDecOpen (); if (faad->handle == NULL) { GST_WARNING_OBJECT (faad, "faacDecOpen() failed"); return FALSE; } conf = faacDecGetCurrentConfiguration (faad->handle); conf->defObjectType = LC; conf->dontUpSampleImplicitSBR = 1; conf->outputFormat = FAAD_FMT_16BIT; if (faacDecSetConfiguration (faad->handle, conf) == 0) { GST_WARNING_OBJECT (faad, "faacDecSetConfiguration() failed"); return FALSE; } return TRUE; } static void gst_faad_close_decoder (GstFaad * faad) { if (faad->handle) { faacDecClose (faad->handle); faad->handle = NULL; } } static gboolean plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, faad, "Free AAC Decoder (FAAD)", plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)