+++ /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_static_metadata (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 / gst-plugins-ugly.git / commitdiff