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851 lines
23 KiB
C
851 lines
23 KiB
C
/* GStreamer FAAD (Free AAC Decoder) plugin
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* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-faad
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* @title: faad
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* @seealso: faac
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*
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* faad decodes AAC (MPEG-4 part 3) stream.
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*
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* ## Example launch lines
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* |[
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* gst-launch-1.0 filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Play aac from mp4 file.
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* |[
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* gst-launch-1.0 filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Play standalone aac bitstream.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/audio/audio.h>
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#include "gstfaad.h"
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GST_DEBUG_CATEGORY_STATIC (faad_debug);
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#define GST_CAT_DEFAULT faad_debug
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 2; "
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"audio/mpeg, mpegversion = (int) 4, stream-format = (string) { raw, adts }")
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);
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#define STATIC_RAW_CAPS(format) \
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"audio/x-raw, " \
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"format = (string) "GST_AUDIO_NE(format)", " \
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"layout = (string) interleaved, " \
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"rate = (int) [ 8000, 96000 ], " \
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"channels = (int) [ 1, 8 ]"
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/*
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* All except 16-bit integer are disabled until someone fixes FAAD.
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* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
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* for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
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* audio, but not for any other. You'll get random segfaults, crashes
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* and even valgrind goes crazy.
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*/
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#define STATIC_CAPS \
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STATIC_RAW_CAPS (S16)
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#if 0
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#define NOTUSED "; " \
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STATIC_RAW_CAPS (S24) \
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"; " \
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STATIC_RAW_CAPS (S32) \
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"; " \
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STATIC_RAW_CAPS (F32) \
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"; " \
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STATIC_RAW_CAPS (F64)
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#endif
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (STATIC_CAPS)
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);
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static void gst_faad_reset (GstFaad * faad);
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static gboolean gst_faad_start (GstAudioDecoder * dec);
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static gboolean gst_faad_stop (GstAudioDecoder * dec);
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static gboolean gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static GstFlowReturn gst_faad_parse (GstAudioDecoder * dec,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_faad_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard);
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static gboolean gst_faad_open_decoder (GstFaad * faad);
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static void gst_faad_close_decoder (GstFaad * faad);
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#define gst_faad_parent_class parent_class
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G_DEFINE_TYPE (GstFaad, gst_faad, GST_TYPE_AUDIO_DECODER);
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static void
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gst_faad_class_init (GstFaadClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_set_static_metadata (element_class, "AAC audio decoder",
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"Codec/Decoder/Audio",
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"Free MPEG-2/4 AAC decoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush);
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GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
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}
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static void
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gst_faad_init (GstFaad * faad)
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{
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(faad), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (faad));
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gst_faad_reset (faad);
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}
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static void
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gst_faad_reset_stream_state (GstFaad * faad)
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{
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if (faad->handle)
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faacDecPostSeekReset (faad->handle, 0);
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}
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static void
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gst_faad_reset (GstFaad * faad)
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{
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faad->samplerate = -1;
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faad->channels = -1;
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faad->init = FALSE;
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faad->packetised = FALSE;
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g_free (faad->channel_positions);
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faad->channel_positions = NULL;
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faad->last_header = 0;
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gst_faad_reset_stream_state (faad);
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}
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static gboolean
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gst_faad_start (GstAudioDecoder * dec)
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{
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GstFaad *faad = GST_FAAD (dec);
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GST_DEBUG_OBJECT (dec, "start");
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gst_faad_reset (faad);
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/* call upon legacy upstream byte support (e.g. seeking) */
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gst_audio_decoder_set_estimate_rate (dec, TRUE);
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/* never mind a few errors */
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gst_audio_decoder_set_max_errors (dec, 10);
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return TRUE;
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}
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static gboolean
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gst_faad_stop (GstAudioDecoder * dec)
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{
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GstFaad *faad = GST_FAAD (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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gst_faad_reset (faad);
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gst_faad_close_decoder (faad);
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return TRUE;
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}
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static gint
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aac_rate_idx (gint rate)
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{
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if (92017 <= rate)
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return 0;
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else if (75132 <= rate)
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return 1;
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else if (55426 <= rate)
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return 2;
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else if (46009 <= rate)
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return 3;
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else if (37566 <= rate)
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return 4;
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else if (27713 <= rate)
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return 5;
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else if (23004 <= rate)
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return 6;
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else if (18783 <= rate)
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return 7;
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else if (13856 <= rate)
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return 8;
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else if (11502 <= rate)
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return 9;
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else if (9391 <= rate)
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return 10;
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else
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return 11;
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}
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static gboolean
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gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstFaad *faad = GST_FAAD (dec);
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GstStructure *str = gst_caps_get_structure (caps, 0);
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GstBuffer *buf;
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const GValue *value;
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GstMapInfo map;
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guint8 *cdata;
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gsize csize;
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/* clean up current decoder, rather than trying to reconfigure */
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gst_faad_close_decoder (faad);
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/* Assume raw stream */
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faad->packetised = FALSE;
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if ((value = gst_structure_get_value (str, "codec_data"))) {
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unsigned long samplerate;
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guint8 channels;
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/* We have codec data, means packetised stream */
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faad->packetised = TRUE;
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buf = gst_value_get_buffer (value);
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g_return_val_if_fail (buf != NULL, FALSE);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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cdata = map.data;
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csize = map.size;
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if (csize < 2)
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goto wrong_length;
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GST_DEBUG_OBJECT (faad,
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"codec_data: object_type=%d, sample_rate=%d, channels=%d",
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((cdata[0] & 0xf8) >> 3),
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(((cdata[0] & 0x07) << 1) | ((cdata[1] & 0x80) >> 7)),
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((cdata[1] & 0x78) >> 3));
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if (!gst_faad_open_decoder (faad))
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goto open_failed;
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/* someone forgot that char can be unsigned when writing the API */
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if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate,
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&channels) < 0)
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goto init_failed;
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if (channels != ((cdata[1] & 0x78) >> 3)) {
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/* https://bugs.launchpad.net/ubuntu/+source/faad2/+bug/290259 */
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GST_WARNING_OBJECT (faad,
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"buggy faad version, wrong nr of channels %d instead of %d", channels,
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((cdata[1] & 0x78) >> 3));
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}
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GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels,
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(guint32) samplerate);
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/* not updating these here, so they are updated in the
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* chain function, and new caps are created etc. */
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faad->samplerate = 0;
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faad->channels = 0;
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faad->init = TRUE;
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gst_buffer_unmap (buf, &map);
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} else if ((value = gst_structure_get_value (str, "framed")) &&
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g_value_get_boolean (value) == TRUE) {
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faad->packetised = TRUE;
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faad->init = FALSE;
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GST_DEBUG_OBJECT (faad, "we have packetized audio");
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} else {
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faad->init = FALSE;
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}
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faad->fake_codec_data[0] = 0;
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faad->fake_codec_data[1] = 0;
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if (faad->packetised && !faad->init) {
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gint rate, channels;
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if (gst_structure_get_int (str, "rate", &rate) &&
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gst_structure_get_int (str, "channels", &channels)) {
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gint rate_idx, profile;
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profile = 3; /* 0=MAIN, 1=LC, 2=SSR, 3=LTP */
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rate_idx = aac_rate_idx (rate);
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faad->fake_codec_data[0] = ((profile + 1) << 3) | ((rate_idx & 0xE) >> 1);
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faad->fake_codec_data[1] = ((rate_idx & 0x1) << 7) | (channels << 3);
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GST_LOG_OBJECT (faad, "created fake codec data (%u,%u): 0x%x 0x%x", rate,
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channels, (int) faad->fake_codec_data[0],
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(int) faad->fake_codec_data[1]);
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}
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}
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return TRUE;
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/* ERRORS */
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wrong_length:
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{
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GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
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gst_buffer_unmap (buf, &map);
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return FALSE;
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}
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open_failed:
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{
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GST_DEBUG_OBJECT (faad, "failed to create decoder");
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gst_buffer_unmap (buf, &map);
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return FALSE;
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}
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init_failed:
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{
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GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
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gst_buffer_unmap (buf, &map);
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return FALSE;
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}
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}
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static gboolean
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gst_faad_chanpos_to_gst (GstFaad * faad, guchar * fpos,
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GstAudioChannelPosition * pos, guint num)
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{
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guint n;
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gboolean unknown_channel = FALSE;
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/* special handling for the common cases for mono and stereo */
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if (num == 1 && fpos[0] == FRONT_CHANNEL_CENTER) {
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GST_DEBUG_OBJECT (faad, "mono common case; won't set channel positions");
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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return TRUE;
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} else if (num == 2 && fpos[0] == FRONT_CHANNEL_LEFT
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&& fpos[1] == FRONT_CHANNEL_RIGHT) {
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GST_DEBUG_OBJECT (faad, "stereo common case; won't set channel positions");
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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return TRUE;
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}
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for (n = 0; n < num; n++) {
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GST_DEBUG_OBJECT (faad, "faad channel %d as %d", n, fpos[n]);
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switch (fpos[n]) {
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case FRONT_CHANNEL_LEFT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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break;
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case FRONT_CHANNEL_RIGHT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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break;
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case FRONT_CHANNEL_CENTER:
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/* argh, mono = center */
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if (num == 1)
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pos[n] = GST_AUDIO_CHANNEL_POSITION_MONO;
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else
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pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
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break;
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case SIDE_CHANNEL_LEFT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
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break;
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case SIDE_CHANNEL_RIGHT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
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break;
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case BACK_CHANNEL_LEFT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
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break;
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case BACK_CHANNEL_RIGHT:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
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break;
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case BACK_CHANNEL_CENTER:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
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break;
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case LFE_CHANNEL:
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pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE1;
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break;
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default:
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GST_DEBUG_OBJECT (faad, "unknown channel %d at %d", fpos[n], n);
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unknown_channel = TRUE;
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break;
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}
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}
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if (unknown_channel) {
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switch (num) {
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case 1:{
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GST_DEBUG_OBJECT (faad,
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"FAAD reports unknown 1 channel mapping. Forcing to mono");
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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break;
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}
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case 2:{
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GST_DEBUG_OBJECT (faad,
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"FAAD reports unknown 2 channel mapping. Forcing to stereo");
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
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pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
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break;
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}
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default:{
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GST_WARNING_OBJECT (faad,
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"Unsupported FAAD channel position 0x%x encountered", fpos[n]);
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return FALSE;
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}
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}
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}
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return TRUE;
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}
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static gboolean
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gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info)
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{
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gboolean ret;
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gboolean fmt_change = FALSE;
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GstAudioInfo ainfo;
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gint i;
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GstAudioChannelPosition position[6];
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/* see if we need to renegotiate */
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if (info->samplerate != faad->samplerate ||
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info->channels != faad->channels || !faad->channel_positions) {
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fmt_change = TRUE;
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} else {
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for (i = 0; i < info->channels; i++) {
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if (info->channel_position[i] != faad->channel_positions[i]) {
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fmt_change = TRUE;
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break;
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}
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}
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}
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if (G_LIKELY (gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (faad))
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&& !fmt_change))
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return TRUE;
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/* store new negotiation information */
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faad->samplerate = info->samplerate;
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faad->channels = info->channels;
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g_free (faad->channel_positions);
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faad->channel_positions = g_memdup (info->channel_position, faad->channels);
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faad->bps = 16 / 8;
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if (!gst_faad_chanpos_to_gst (faad, faad->channel_positions,
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faad->aac_positions, faad->channels)) {
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GST_DEBUG_OBJECT (faad, "Could not map channel positions");
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return FALSE;
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}
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memcpy (position, faad->aac_positions, sizeof (position));
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gst_audio_channel_positions_to_valid_order (position, faad->channels);
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memcpy (faad->gst_positions, position,
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faad->channels * sizeof (GstAudioChannelPosition));
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/* get the remap table */
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memset (faad->reorder_map, 0, sizeof (faad->reorder_map));
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faad->need_reorder = FALSE;
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if (gst_audio_get_channel_reorder_map (faad->channels, faad->aac_positions,
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faad->gst_positions, faad->reorder_map)) {
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for (i = 0; i < faad->channels; i++) {
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GST_DEBUG_OBJECT (faad, "remap %d -> %d", i, faad->reorder_map[i]);
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if (faad->reorder_map[i] != i) {
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faad->need_reorder = TRUE;
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}
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}
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}
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|
|
/* 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, position);
|
|
|
|
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) {
|
|
unsigned long rate;
|
|
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_SECONDARY,
|
|
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)
|