--- /dev/null
--- /dev/null
++/* GStreamer
++ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
++ *
++ * This library is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Library General Public
++ * License as published by the Free Software Foundation; either
++ * version 2 of the License, or (at your option) any later version.
++ *
++ * This library is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
++ * Library General Public License for more details.
++ *
++ * You should have received a copy of the GNU Library General Public
++ * License along with this library; if not, write to the
++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
++ * Boston, MA 02111-1307, USA.
++ */
++
++/**
++ * SECTION:element-mad
++ * @see_also: lame
++ *
++ * MP3 audio decoder.
++ *
++ * <refsect2>
++ * <title>Example pipelines</title>
++ * |[
++ * gst-launch filesrc location=music.mp3 ! mpegaudioparse ! mad ! audioconvert ! audioresample ! autoaudiosink
++ * ]| Decode and play the mp3 file
++ * </refsect2>
++ */
++
++#ifdef HAVE_CONFIG_H
++#include "config.h"
++#endif
++
++#include <stdlib.h>
++#include <string.h>
++#include "gstmad.h"
++#include <gst/audio/audio.h>
++
++enum
++{
++ ARG_0,
++ ARG_HALF,
++ ARG_IGNORE_CRC
++};
++
++GST_DEBUG_CATEGORY_STATIC (mad_debug);
++#define GST_CAT_DEFAULT mad_debug
++
++static GstStaticPadTemplate mad_src_template_factory =
++GST_STATIC_PAD_TEMPLATE ("src",
++ GST_PAD_SRC,
++ GST_PAD_ALWAYS,
++ GST_STATIC_CAPS ("audio/x-raw, "
++ "format = (string) " GST_AUDIO_NE (S32) ", "
++ "layout = (string) interleaved, "
++ "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
++ "channels = (int) [ 1, 2 ]")
++ );
++
++/* FIXME: make three caps, for mpegversion 1, 2 and 2.5 */
++static GstStaticPadTemplate mad_sink_template_factory =
++GST_STATIC_PAD_TEMPLATE ("sink",
++ GST_PAD_SINK,
++ GST_PAD_ALWAYS,
++ GST_STATIC_CAPS ("audio/mpeg, "
++ "mpegversion = (int) 1, "
++ "layer = (int) [ 1, 3 ], "
++ "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
++ "channels = (int) [ 1, 2 ]")
++ );
++
++
++static gboolean gst_mad_start (GstAudioDecoder * dec);
++static gboolean gst_mad_stop (GstAudioDecoder * dec);
++static gboolean gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
++ gint * offset, gint * length);
++static GstFlowReturn gst_mad_handle_frame (GstAudioDecoder * dec,
++ GstBuffer * buffer);
++static void gst_mad_flush (GstAudioDecoder * dec, gboolean hard);
++
++static void gst_mad_set_property (GObject * object, guint prop_id,
++ const GValue * value, GParamSpec * pspec);
++static void gst_mad_get_property (GObject * object, guint prop_id,
++ GValue * value, GParamSpec * pspec);
++
++#define parent_class gst_mad_parent_class
++G_DEFINE_TYPE (GstMad, gst_mad, GST_TYPE_AUDIO_DECODER);
++
++static void
++gst_mad_class_init (GstMadClass * klass)
++{
++ GObjectClass *gobject_class = (GObjectClass *) klass;
++ GstElementClass *element_class = (GstElementClass *) klass;
++ GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
++
++ base_class->start = GST_DEBUG_FUNCPTR (gst_mad_start);
++ base_class->stop = GST_DEBUG_FUNCPTR (gst_mad_stop);
++ base_class->parse = GST_DEBUG_FUNCPTR (gst_mad_parse);
++ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
++ base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
++
++ base_class->start = GST_DEBUG_FUNCPTR (gst_mad_start);
++ base_class->stop = GST_DEBUG_FUNCPTR (gst_mad_stop);
++ base_class->parse = GST_DEBUG_FUNCPTR (gst_mad_parse);
++ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
++ base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
++
++ gobject_class->set_property = gst_mad_set_property;
++ gobject_class->get_property = gst_mad_get_property;
++
++ /* init properties */
++ /* currently, string representations are used, we might want to change that */
++ /* FIXME: descriptions need to be more technical,
++ * default values and ranges need to be selected right */
++ g_object_class_install_property (gobject_class, ARG_HALF,
++ g_param_spec_boolean ("half", "Half", "Generate PCM at 1/2 sample rate",
++ FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++ g_object_class_install_property (gobject_class, ARG_IGNORE_CRC,
++ g_param_spec_boolean ("ignore-crc", "Ignore CRC", "Ignore CRC errors",
++ TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
++
++ gst_element_class_add_pad_template (element_class,
++ gst_static_pad_template_get (&mad_sink_template_factory));
++ gst_element_class_add_pad_template (element_class,
++ gst_static_pad_template_get (&mad_src_template_factory));
++
++ gst_element_class_set_details_simple (element_class, "mad mp3 decoder",
++ "Codec/Decoder/Audio",
++ "Uses mad code to decode mp3 streams", "Wim Taymans <wim@fluendo.com>");
++}
++
++static void
++gst_mad_init (GstMad * mad)
++{
++ GstAudioDecoder *dec;
++
++ dec = GST_AUDIO_DECODER (mad);
++ gst_audio_decoder_set_tolerance (dec, 20 * GST_MSECOND);
++
++ mad->half = FALSE;
++ mad->ignore_crc = TRUE;
++}
++
++static gboolean
++gst_mad_start (GstAudioDecoder * dec)
++{
++ GstMad *mad = GST_MAD (dec);
++ guint options = 0;
++
++ GST_DEBUG_OBJECT (dec, "start");
++ mad_stream_init (&mad->stream);
++ mad_frame_init (&mad->frame);
++ mad_synth_init (&mad->synth);
++ mad->rate = 0;
++ mad->channels = 0;
++ mad->caps_set = FALSE;
++ mad->frame.header.samplerate = 0;
++ if (mad->ignore_crc)
++ options |= MAD_OPTION_IGNORECRC;
++ if (mad->half)
++ options |= MAD_OPTION_HALFSAMPLERATE;
++ mad_stream_options (&mad->stream, options);
++ mad->header.mode = -1;
++ mad->header.emphasis = -1;
++ mad->eos = FALSE;
++
++ /* call upon legacy upstream byte support (e.g. seeking) */
++ gst_audio_decoder_set_byte_time (dec, TRUE);
++
++ return TRUE;
++}
++
++static gboolean
++gst_mad_stop (GstAudioDecoder * dec)
++{
++ GstMad *mad = GST_MAD (dec);
++
++ GST_DEBUG_OBJECT (dec, "stop");
++ mad_synth_finish (&mad->synth);
++ mad_frame_finish (&mad->frame);
++ mad_stream_finish (&mad->stream);
++
++ return TRUE;
++}
++
++static inline gint32
++scale (mad_fixed_t sample)
++{
++#if MAD_F_FRACBITS < 28
++ /* round */
++ sample += (1L << (28 - MAD_F_FRACBITS - 1));
++#endif
++
++ /* clip */
++ if (sample >= MAD_F_ONE)
++ sample = MAD_F_ONE - 1;
++ else if (sample < -MAD_F_ONE)
++ sample = -MAD_F_ONE;
++
++#if MAD_F_FRACBITS < 28
++ /* quantize */
++ sample >>= (28 - MAD_F_FRACBITS);
++#endif
++
++ /* convert from 29 bits to 32 bits */
++ return (gint32) (sample << 3);
++}
++
++/* internal function to check if the header has changed and thus the
++ * caps need to be reset. Only call during normal mode, not resyncing */
++static void
++gst_mad_check_caps_reset (GstMad * mad)
++{
++ guint nchannels;
++ guint rate;
++
++ nchannels = MAD_NCHANNELS (&mad->frame.header);
++
++#if MAD_VERSION_MINOR <= 12
++ rate = mad->header.sfreq;
++#else
++ rate = mad->frame.header.samplerate;
++#endif
++
++ /* rate and channels are not supposed to change in a continuous stream,
++ * so check this first before doing anything */
++
++ /* only set caps if they weren't already set for this continuous stream */
++ if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (mad))
++ || mad->channels != nchannels || mad->rate != rate) {
++ GstAudioInfo info;
++ static const GstAudioChannelPosition chan_pos[2][2] = {
++ {GST_AUDIO_CHANNEL_POSITION_MONO},
++ {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
++ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}
++ };
++
++ if (mad->caps_set) {
++ GST_DEBUG_OBJECT (mad, "Header changed from %d Hz/%d ch to %d Hz/%d ch, "
++ "failed sync after seek ?", mad->rate, mad->channels, rate,
++ nchannels);
++ /* we're conservative on stream changes. However, our *initial* caps
++ * might have been wrong as well - mad ain't perfect in syncing. So,
++ * we count caps changes and change if we pass a limit treshold (3). */
++ if (nchannels != mad->pending_channels || rate != mad->pending_rate) {
++ mad->times_pending = 0;
++ mad->pending_channels = nchannels;
++ mad->pending_rate = rate;
++ }
++ if (++mad->times_pending < 3)
++ return;
++ }
++
++ if (mad->stream.options & MAD_OPTION_HALFSAMPLERATE)
++ rate >>= 1;
++
++ /* we set the caps even when the pad is not connected so they
++ * can be gotten for streaminfo */
++ gst_audio_info_init (&info);
++ gst_audio_info_set_format (&info,
++ GST_AUDIO_FORMAT_S32, rate, nchannels, chan_pos[nchannels - 1]);
++
++ gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (mad), &info);
++
++ mad->caps_set = TRUE;
++ mad->channels = nchannels;
++ mad->rate = rate;
++ }
++}
++
++static GstFlowReturn
++gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
++ gint * _offset, gint * len)
++{
++ GstMad *mad;
++<<<<<<< HEAD
++ GstFlowReturn ret = GST_FLOW_EOS;
++ gint av, size, offset, prev_offset, consumed = 0;
++ const guint8 *data;
++ gboolean eos;
++ guint8 *guard = NULL;
++=======
++ GstFlowReturn ret = GST_FLOW_UNEXPECTED;
++ gint av, size, offset;
++ const guint8 *data;
++ gboolean eos, sync;
++ GstBuffer *guard = NULL;
++>>>>>>> origin/master
++
++ mad = GST_MAD (dec);
++
++ av = gst_adapter_available (adapter);
++ data = gst_adapter_map (adapter, av);
++
++ gst_audio_decoder_get_parse_state (dec, &sync, &eos);
++ GST_LOG_OBJECT (mad, "parse state sync %d, eos %d", sync, eos);
++
++ if (eos) {
++ /* This is one streaming hack right there.
++ * mad will not decode the last frame if it is not followed by
++ * a number of 0 bytes, due to some buffer overflow, which can
++ * not be fixed for reasons I did not inquire into, see
++ * http://www.mars.org/mailman/public/mad-dev/2001-May/000262.html
++ */
++ guard = g_malloc (av + MAD_BUFFER_GUARD);
++ /* let's be nice and not mess with baseclass state and keep hacks local */
++ memcpy (guard, data, av);
++ memset (guard + av, 0, MAD_BUFFER_GUARD);
++ GST_DEBUG_OBJECT (mad, "Added %u zero guard bytes in the adapter; "
++ "using fallback buffer of size %u",
++ MAD_BUFFER_GUARD, av + MAD_BUFFER_GUARD);
++ data = guard;
++ av = av + MAD_BUFFER_GUARD;
++ }
++
++ /* we basically let mad library do parsing,
++ * and translate that back to baseclass.
++ * if a frame is found (and also decoded), subsequent handle_frame
++ * only needs to synthesize it */
++
++ offset = 0;
++ size = 0;
++
++resume:
++ if (G_UNLIKELY (offset + MAD_BUFFER_GUARD > av))
++ goto exit;
++
++ GST_LOG_OBJECT (mad, "setup mad stream at offset %d (of av %d)", offset, av);
++ mad_stream_buffer (&mad->stream, data + offset, av - offset);
++ /* convey sync idea to mad */
++ mad->stream.sync = sync;
++ /* if we get back here, lost sync anyway */
++ sync = FALSE;
++
++ while (TRUE) {
++ GST_LOG_OBJECT (mad, "decoding the header now");
++ if (mad_header_decode (&mad->frame.header, &mad->stream) == -1) {
++ /* HACK it seems mad reports wrong error when it is trying to determine
++ * free bitrate and scanning for next header */
++ if (mad->stream.error == MAD_ERROR_LOSTSYNC) {
++ const guint8 *ptr = mad->stream.this_frame;
++ guint32 header;
++
++ if (ptr >= data && ptr < data + av) {
++ header = GST_READ_UINT32_BE (ptr);
++ /* looks like possible freeform header with not much data */
++ if (((header & 0xFFE00000) == 0xFFE00000) &&
++ (((header >> 12) & 0xF) == 0x0) && (av < 4096)) {
++ GST_DEBUG_OBJECT (mad, "overriding freeform LOST_SYNC to BUFLEN");
++ mad->stream.error = MAD_ERROR_BUFLEN;
++ }
++ }
++ }
++ if (mad->stream.error == MAD_ERROR_BUFLEN) {
++ GST_LOG_OBJECT (mad, "not enough data, getting more");
++ 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);
projects / platform / upstream / gstreamer.git / commitdiff