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mad: remove .c.orig file
This commit is contained in:
Tim-Philipp Müller
parent
08d20e4da8
commit
3f5ff831fc
1 changed files with 0 additions and 567 deletions
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@ -1,567 +0,0 @@
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-mad
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* @see_also: lame
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*
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* MP3 audio decoder.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch filesrc location=music.mp3 ! mpegaudioparse ! mad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Decode and play the mp3 file
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* </refsect2>
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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 <stdlib.h>
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#include <string.h>
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#include "gstmad.h"
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#include <gst/audio/audio.h>
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enum
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{
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ARG_0,
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ARG_HALF,
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ARG_IGNORE_CRC
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};
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GST_DEBUG_CATEGORY_STATIC (mad_debug);
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#define GST_CAT_DEFAULT mad_debug
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static GstStaticPadTemplate mad_src_template_factory =
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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 ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S32) ", "
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"layout = (string) interleaved, "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) [ 1, 2 ]")
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);
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/* FIXME: make three caps, for mpegversion 1, 2 and 2.5 */
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static GstStaticPadTemplate mad_sink_template_factory =
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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, "
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"mpegversion = (int) 1, "
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"layer = (int) [ 1, 3 ], "
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
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"channels = (int) [ 1, 2 ]")
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);
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static gboolean gst_mad_start (GstAudioDecoder * dec);
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static gboolean gst_mad_stop (GstAudioDecoder * dec);
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static gboolean gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_mad_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void gst_mad_flush (GstAudioDecoder * dec, gboolean hard);
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static void gst_mad_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_mad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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#define parent_class gst_mad_parent_class
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G_DEFINE_TYPE (GstMad, gst_mad, GST_TYPE_AUDIO_DECODER);
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static void
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gst_mad_class_init (GstMadClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) klass;
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GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
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base_class->start = GST_DEBUG_FUNCPTR (gst_mad_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_mad_stop);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_mad_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
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base_class->start = GST_DEBUG_FUNCPTR (gst_mad_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_mad_stop);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_mad_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
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gobject_class->set_property = gst_mad_set_property;
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gobject_class->get_property = gst_mad_get_property;
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/* init properties */
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/* currently, string representations are used, we might want to change that */
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/* FIXME: descriptions need to be more technical,
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* default values and ranges need to be selected right */
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g_object_class_install_property (gobject_class, ARG_HALF,
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g_param_spec_boolean ("half", "Half", "Generate PCM at 1/2 sample rate",
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FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_IGNORE_CRC,
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g_param_spec_boolean ("ignore-crc", "Ignore CRC", "Ignore CRC errors",
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TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&mad_sink_template_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&mad_src_template_factory));
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gst_element_class_set_static_metadata (element_class, "mad mp3 decoder",
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"Codec/Decoder/Audio",
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"Uses mad code to decode mp3 streams", "Wim Taymans <wim@fluendo.com>");
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}
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static void
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gst_mad_init (GstMad * mad)
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{
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GstAudioDecoder *dec;
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dec = GST_AUDIO_DECODER (mad);
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gst_audio_decoder_set_tolerance (dec, 20 * GST_MSECOND);
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mad->half = FALSE;
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mad->ignore_crc = TRUE;
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}
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static gboolean
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gst_mad_start (GstAudioDecoder * dec)
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{
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GstMad *mad = GST_MAD (dec);
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guint options = 0;
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GST_DEBUG_OBJECT (dec, "start");
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mad_stream_init (&mad->stream);
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mad_frame_init (&mad->frame);
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mad_synth_init (&mad->synth);
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mad->rate = 0;
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mad->channels = 0;
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mad->caps_set = FALSE;
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mad->frame.header.samplerate = 0;
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if (mad->ignore_crc)
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options |= MAD_OPTION_IGNORECRC;
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if (mad->half)
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options |= MAD_OPTION_HALFSAMPLERATE;
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mad_stream_options (&mad->stream, options);
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mad->header.mode = -1;
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mad->header.emphasis = -1;
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mad->eos = FALSE;
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/* call upon legacy upstream byte support (e.g. seeking) */
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gst_audio_decoder_set_byte_time (dec, TRUE);
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return TRUE;
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}
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static gboolean
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gst_mad_stop (GstAudioDecoder * dec)
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{
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GstMad *mad = GST_MAD (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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mad_synth_finish (&mad->synth);
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mad_frame_finish (&mad->frame);
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mad_stream_finish (&mad->stream);
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return TRUE;
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}
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static inline gint32
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scale (mad_fixed_t sample)
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{
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#if MAD_F_FRACBITS < 28
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/* round */
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sample += (1L << (28 - MAD_F_FRACBITS - 1));
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#endif
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/* clip */
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if (sample >= MAD_F_ONE)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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#if MAD_F_FRACBITS < 28
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/* quantize */
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sample >>= (28 - MAD_F_FRACBITS);
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#endif
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/* convert from 29 bits to 32 bits */
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return (gint32) (sample << 3);
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}
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/* internal function to check if the header has changed and thus the
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* caps need to be reset. Only call during normal mode, not resyncing */
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static void
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gst_mad_check_caps_reset (GstMad * mad)
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{
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guint nchannels;
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guint rate;
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nchannels = MAD_NCHANNELS (&mad->frame.header);
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#if MAD_VERSION_MINOR <= 12
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rate = mad->header.sfreq;
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#else
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rate = mad->frame.header.samplerate;
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#endif
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/* rate and channels are not supposed to change in a continuous stream,
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* so check this first before doing anything */
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/* only set caps if they weren't already set for this continuous stream */
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if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (mad))
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|| mad->channels != nchannels || mad->rate != rate) {
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GstAudioInfo info;
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static const GstAudioChannelPosition chan_pos[2][2] = {
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{GST_AUDIO_CHANNEL_POSITION_MONO},
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{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}
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};
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if (mad->caps_set) {
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GST_DEBUG_OBJECT (mad, "Header changed from %d Hz/%d ch to %d Hz/%d ch, "
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"failed sync after seek ?", mad->rate, mad->channels, rate,
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nchannels);
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/* we're conservative on stream changes. However, our *initial* caps
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* might have been wrong as well - mad ain't perfect in syncing. So,
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* we count caps changes and change if we pass a limit treshold (3). */
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if (nchannels != mad->pending_channels || rate != mad->pending_rate) {
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mad->times_pending = 0;
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mad->pending_channels = nchannels;
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mad->pending_rate = rate;
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}
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if (++mad->times_pending < 3)
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return;
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}
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if (mad->stream.options & MAD_OPTION_HALFSAMPLERATE)
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rate >>= 1;
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/* we set the caps even when the pad is not connected so they
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* can be gotten for streaminfo */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info,
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GST_AUDIO_FORMAT_S32, rate, nchannels, chan_pos[nchannels - 1]);
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (mad), &info);
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mad->caps_set = TRUE;
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mad->channels = nchannels;
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mad->rate = rate;
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}
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}
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static GstFlowReturn
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gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * _offset, gint * len)
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{
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GstMad *mad;
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<<<<<<< HEAD
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GstFlowReturn ret = GST_FLOW_EOS;
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gint av, size, offset, prev_offset, consumed = 0;
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const guint8 *data;
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gboolean eos;
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guint8 *guard = NULL;
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=======
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GstFlowReturn ret = GST_FLOW_UNEXPECTED;
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gint av, size, offset;
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const guint8 *data;
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gboolean eos, sync;
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GstBuffer *guard = NULL;
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>>>>>>> origin/master
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mad = GST_MAD (dec);
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av = gst_adapter_available (adapter);
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data = gst_adapter_map (adapter, av);
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gst_audio_decoder_get_parse_state (dec, &sync, &eos);
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GST_LOG_OBJECT (mad, "parse state sync %d, eos %d", sync, eos);
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if (eos) {
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/* This is one streaming hack right there.
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* mad will not decode the last frame if it is not followed by
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* a number of 0 bytes, due to some buffer overflow, which can
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* not be fixed for reasons I did not inquire into, see
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* http://www.mars.org/mailman/public/mad-dev/2001-May/000262.html
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*/
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guard = g_malloc (av + MAD_BUFFER_GUARD);
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/* let's be nice and not mess with baseclass state and keep hacks local */
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memcpy (guard, data, av);
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memset (guard + av, 0, MAD_BUFFER_GUARD);
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GST_DEBUG_OBJECT (mad, "Added %u zero guard bytes in the adapter; "
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"using fallback buffer of size %u",
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MAD_BUFFER_GUARD, av + MAD_BUFFER_GUARD);
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data = guard;
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av = av + MAD_BUFFER_GUARD;
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}
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/* we basically let mad library do parsing,
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* and translate that back to baseclass.
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* if a frame is found (and also decoded), subsequent handle_frame
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* only needs to synthesize it */
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offset = 0;
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size = 0;
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resume:
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if (G_UNLIKELY (offset + MAD_BUFFER_GUARD > av))
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goto exit;
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GST_LOG_OBJECT (mad, "setup mad stream at offset %d (of av %d)", offset, av);
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mad_stream_buffer (&mad->stream, data + offset, av - offset);
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/* convey sync idea to mad */
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mad->stream.sync = sync;
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/* if we get back here, lost sync anyway */
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sync = FALSE;
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while (TRUE) {
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GST_LOG_OBJECT (mad, "decoding the header now");
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if (mad_header_decode (&mad->frame.header, &mad->stream) == -1) {
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/* HACK it seems mad reports wrong error when it is trying to determine
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* free bitrate and scanning for next header */
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if (mad->stream.error == MAD_ERROR_LOSTSYNC) {
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const guint8 *ptr = mad->stream.this_frame;
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guint32 header;
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if (ptr >= data && ptr < data + av) {
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header = GST_READ_UINT32_BE (ptr);
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/* looks like possible freeform header with not much data */
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if (((header & 0xFFE00000) == 0xFFE00000) &&
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(((header >> 12) & 0xF) == 0x0) && (av < 4096)) {
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GST_DEBUG_OBJECT (mad, "overriding freeform LOST_SYNC to BUFLEN");
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mad->stream.error = MAD_ERROR_BUFLEN;
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}
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}
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}
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if (mad->stream.error == MAD_ERROR_BUFLEN) {
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GST_LOG_OBJECT (mad, "not enough data, getting more");
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offset = mad->stream.next_frame - data;
|
|
||||||
break;
|
|
||||||
} else if (mad->stream.error == MAD_ERROR_LOSTSYNC) {
|
|
||||||
GST_LOG_OBJECT (mad, "lost sync");
|
|
||||||
continue;
|
|
||||||
} else {
|
|
||||||
/* probably some bogus header, basically also lost sync */
|
|
||||||
GST_DEBUG_OBJECT (mad, "mad_header_decode had an error: %s",
|
|
||||||
mad_stream_errorstr (&mad->stream));
|
|
||||||
continue;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* could have a frame now, subsequent will confirm */
|
|
||||||
offset = mad->stream.this_frame - data;
|
|
||||||
size = mad->stream.next_frame - mad->stream.this_frame;
|
|
||||||
g_assert (size);
|
|
||||||
|
|
||||||
GST_LOG_OBJECT (mad, "parsing and decoding one frame now "
|
|
||||||
"(offset %d, size %d)", offset, size);
|
|
||||||
if (mad_frame_decode (&mad->frame, &mad->stream) == -1) {
|
|
||||||
GST_LOG_OBJECT (mad, "got error %d", mad->stream.error);
|
|
||||||
|
|
||||||
/* not enough data, need to wait for next buffer? */
|
|
||||||
if (mad->stream.error == MAD_ERROR_BUFLEN) {
|
|
||||||
/* not really expect this error at this stage anymore
|
|
||||||
* assume bogus frame and bad sync and move along a bit */
|
|
||||||
GST_WARNING_OBJECT (mad, "not enough data (unexpected), moving along");
|
|
||||||
offset++;
|
|
||||||
goto resume;
|
|
||||||
} else if (mad->stream.error == MAD_ERROR_BADDATAPTR) {
|
|
||||||
GST_DEBUG_OBJECT (mad, "bad data ptr, skipping presumed frame");
|
|
||||||
/* flush past presumed frame */
|
|
||||||
offset += size;
|
|
||||||
goto resume;
|
|
||||||
} else {
|
|
||||||
GST_WARNING_OBJECT (mad, "mad_frame_decode had an error: %s",
|
|
||||||
mad_stream_errorstr (&mad->stream));
|
|
||||||
if (!MAD_RECOVERABLE (mad->stream.error)) {
|
|
||||||
/* well, all may be well enough bytes later on ... */
|
|
||||||
GST_AUDIO_DECODER_ERROR (mad, 1, STREAM, DECODE, (NULL),
|
|
||||||
("mad error: %s", mad_stream_errorstr (&mad->stream)), ret);
|
|
||||||
}
|
|
||||||
/* move along and try again */
|
|
||||||
offset++;
|
|
||||||
goto resume;
|
|
||||||
}
|
|
||||||
g_assert_not_reached ();
|
|
||||||
}
|
|
||||||
|
|
||||||
/* so decoded ok, got a frame now */
|
|
||||||
ret = GST_FLOW_OK;
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
|
|
||||||
exit:
|
|
||||||
|
|
||||||
gst_adapter_unmap (adapter);
|
|
||||||
|
|
||||||
*_offset = offset;
|
|
||||||
*len = size;
|
|
||||||
|
|
||||||
/* ensure that if we added some dummy guard bytes above, we don't claim
|
|
||||||
to have used them as they're unknown to the caller. */
|
|
||||||
if (eos) {
|
|
||||||
g_assert (av >= MAD_BUFFER_GUARD);
|
|
||||||
av -= MAD_BUFFER_GUARD;
|
|
||||||
if (*_offset > av)
|
|
||||||
*_offset = av;
|
|
||||||
if (*len > av)
|
|
||||||
*len = av;
|
|
||||||
g_assert (guard);
|
|
||||||
g_free (guard);
|
|
||||||
}
|
|
||||||
|
|
||||||
return ret;
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstFlowReturn
|
|
||||||
gst_mad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
|
|
||||||
{
|
|
||||||
GstMad *mad;
|
|
||||||
GstFlowReturn ret = GST_FLOW_EOS;
|
|
||||||
GstBuffer *outbuffer;
|
|
||||||
guint nsamples;
|
|
||||||
GstMapInfo outmap;
|
|
||||||
gint32 *outdata;
|
|
||||||
mad_fixed_t const *left_ch, *right_ch;
|
|
||||||
|
|
||||||
mad = GST_MAD (dec);
|
|
||||||
|
|
||||||
/* no fancy draining */
|
|
||||||
if (G_UNLIKELY (!buffer))
|
|
||||||
return GST_FLOW_OK;
|
|
||||||
|
|
||||||
/* _parse prepared a frame */
|
|
||||||
nsamples = MAD_NSBSAMPLES (&mad->frame.header) *
|
|
||||||
(mad->stream.options & MAD_OPTION_HALFSAMPLERATE ? 16 : 32);
|
|
||||||
GST_LOG_OBJECT (mad, "mad frame with %d samples", nsamples);
|
|
||||||
|
|
||||||
/* arrange for initial caps before pushing data,
|
|
||||||
* and update later on if needed */
|
|
||||||
gst_mad_check_caps_reset (mad);
|
|
||||||
|
|
||||||
mad_synth_frame (&mad->synth, &mad->frame);
|
|
||||||
left_ch = mad->synth.pcm.samples[0];
|
|
||||||
right_ch = mad->synth.pcm.samples[1];
|
|
||||||
|
|
||||||
outbuffer = gst_buffer_new_and_alloc (nsamples * mad->channels * 4);
|
|
||||||
|
|
||||||
gst_buffer_map (outbuffer, &outmap, GST_MAP_WRITE);
|
|
||||||
outdata = (gint32 *) outmap.data;
|
|
||||||
|
|
||||||
/* output sample(s) in 16-bit signed native-endian PCM */
|
|
||||||
if (mad->channels == 1) {
|
|
||||||
gint count = nsamples;
|
|
||||||
|
|
||||||
while (count--) {
|
|
||||||
*outdata++ = scale (*left_ch++) & 0xffffffff;
|
|
||||||
}
|
|
||||||
} else {
|
|
||||||
gint count = nsamples;
|
|
||||||
|
|
||||||
while (count--) {
|
|
||||||
*outdata++ = scale (*left_ch++) & 0xffffffff;
|
|
||||||
*outdata++ = scale (*right_ch++) & 0xffffffff;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
gst_buffer_unmap (outbuffer, &outmap);
|
|
||||||
|
|
||||||
ret = gst_audio_decoder_finish_frame (dec, outbuffer, 1);
|
|
||||||
|
|
||||||
return ret;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_mad_flush (GstAudioDecoder * dec, gboolean hard)
|
|
||||||
{
|
|
||||||
GstMad *mad;
|
|
||||||
|
|
||||||
mad = GST_MAD (dec);
|
|
||||||
if (hard) {
|
|
||||||
mad_frame_mute (&mad->frame);
|
|
||||||
mad_synth_mute (&mad->synth);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_mad_set_property (GObject * object, guint prop_id,
|
|
||||||
const GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstMad *mad;
|
|
||||||
|
|
||||||
mad = GST_MAD (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case ARG_HALF:
|
|
||||||
mad->half = g_value_get_boolean (value);
|
|
||||||
break;
|
|
||||||
case ARG_IGNORE_CRC:
|
|
||||||
mad->ignore_crc = g_value_get_boolean (value);
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_mad_get_property (GObject * object, guint prop_id,
|
|
||||||
GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstMad *mad;
|
|
||||||
|
|
||||||
mad = GST_MAD (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case ARG_HALF:
|
|
||||||
g_value_set_boolean (value, mad->half);
|
|
||||||
break;
|
|
||||||
case ARG_IGNORE_CRC:
|
|
||||||
g_value_set_boolean (value, mad->ignore_crc);
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* plugin initialisation */
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
plugin_init (GstPlugin * plugin)
|
|
||||||
{
|
|
||||||
GST_DEBUG_CATEGORY_INIT (mad_debug, "mad", 0, "mad mp3 decoding");
|
|
||||||
|
|
||||||
/* FIXME 0.11: rename to something better like madmp3dec or madmpegaudiodec
|
|
||||||
* or so? */
|
|
||||||
return gst_element_register (plugin, "mad", GST_RANK_SECONDARY,
|
|
||||||
gst_mad_get_type ());
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
||||||
GST_VERSION_MINOR,
|
|
||||||
mad,
|
|
||||||
"mp3 decoding based on the mad library",
|
|
||||||
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
|
|
||||||
Loading…
Reference in a new issue