--- /dev/null
+/* MP3 decoding plugin for GStreamer using the mpg123 library
+ * Copyright (C) 2012 Carlos Rafael Giani
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstmpg123audiodec.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+GST_DEBUG_CATEGORY_STATIC (mpg123_debug);
+#define GST_CAT_DEFAULT mpg123_debug
+
+/*
+ * Omitted sample formats that mpg123 supports (or at least can support):
+ * - 8bit integer signed
+ * - 8bit integer unsigned
+ * - a-law
+ * - mu-law
+ * - 64bit float
+ *
+ * The first four formats are not supported by the GstAudioDecoder base class.
+ * (The internal gst_audio_format_from_caps_structure() call fails.)
+ *
+ * The 64bit float issue is tricky. mpg123 actually decodes to "real",
+ * not necessarily to "float".
+ *
+ * "real" can be fixed point, 32bit float, 64bit float. There seems to be
+ * no way how to find out which one of them is actually used.
+ *
+ * However, in all known installations, "real" equals 32bit float, so that's
+ * what is used.
+ */
+
+static GstStaticPadTemplate sink_template = 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 ], " "parsed = (boolean) true ")
+ );
+
+static void gst_mpg123_audio_dec_finalize (GObject * object);
+static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec);
+static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec);
+static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec
+ * mpg123_decoder, unsigned char const *decoded_bytes,
+ size_t const num_decoded_bytes);
+static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
+static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec,
+ GstCaps * incoming_caps);
+static void gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard);
+
+G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER);
+
+static void
+gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
+{
+ GObjectClass *object_class;
+ GstAudioDecoderClass *base_class;
+ GstElementClass *element_class;
+ GstCaps *src_caps;
+ GstPadTemplate *src_template;
+ int error;
+
+ GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
+
+ object_class = G_OBJECT_CLASS (klass);
+ base_class = GST_AUDIO_DECODER_CLASS (klass);
+ element_class = GST_ELEMENT_CLASS (klass);
+
+ object_class->finalize = gst_mpg123_audio_dec_finalize;
+
+ gst_element_class_set_static_metadata (element_class,
+ "mpg123 mp3 decoder",
+ "Codec/Decoder/Audio",
+ "Decodes mp3 streams using the mpg123 library",
+ "Carlos Rafael Giani <dv@pseudoterminal.org>");
+
+ /*
+ Not using static pad template for srccaps, since the comma-separated list of formats needs to be
+ created depending on whatever mpg123 supports
+ */
+ {
+ gchar *format_string;
+ gchar *caps_string;
+
+ static gchar const *src_caps_begin = "audio/x-raw, " "format = { ";
+ static gchar const *src_caps_end =
+ " }, "
+ "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
+ "channels = (int) [ 1, 2 ], " "layout = (string) interleaved; ";
+
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+#define ENDIAN_POSTFIX "LE"
+#else
+#define ENDIAN_POSTFIX "BE"
+#endif
+
+ static gchar *supported_formats[] = {
+#ifdef MPG123_ENC_SIGNED_16
+ "S16" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_UNSIGNED_16
+ "U16" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_SIGNED_24
+ "S24" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_UNSIGNED_24
+ "U24" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_SIGNED_32
+ "S32" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_UNSIGNED_32
+ "U32" ENDIAN_POSTFIX,
+#endif
+#ifdef MPG123_ENC_FLOAT_32
+ "F32" ENDIAN_POSTFIX,
+#endif
+ NULL
+ };
+
+ format_string = g_strjoinv (", ", supported_formats);
+ caps_string =
+ g_strjoin (NULL, src_caps_begin, format_string, src_caps_end, NULL);
+
+ src_caps = gst_caps_from_string (caps_string);
+ src_template =
+ gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+ gst_caps_ref (src_caps));
+
+ g_free (format_string);
+ g_free (caps_string);
+ }
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (element_class, src_template);
+
+ base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop);
+ base_class->handle_frame =
+ GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_handle_frame);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_set_format);
+ base_class->flush = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_flush);
+
+ error = mpg123_init ();
+ if (G_UNLIKELY (error != MPG123_OK))
+ GST_ERROR ("Could not initialize mpg123 library: %s",
+ mpg123_plain_strerror (error));
+ else
+ GST_TRACE ("mpg123 library initialized");
+}
+
+
+void
+gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
+{
+ mpg123_decoder->handle = NULL;
+}
+
+
+static void
+gst_mpg123_audio_dec_finalize (GObject * object)
+{
+ GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (object);
+ if (G_LIKELY (mpg123_decoder->handle != NULL)) {
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ }
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_start (GstAudioDecoder * dec)
+{
+ GstMpg123AudioDec *mpg123_decoder;
+ int error;
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+ error = 0;
+
+ mpg123_decoder->handle = mpg123_new (NULL, &error);
+ mpg123_decoder->has_next_audioinfo = FALSE;
+ mpg123_decoder->frame_offset = 0;
+
+ /*
+ Initially, the mpg123 handle comes with a set of default formats supported. This clears this set.
+ This is necessary, since only one format shall be supported (see set_format for more).
+ */
+ mpg123_format_none (mpg123_decoder->handle);
+
+ mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0); /* Built-in mpg123 support for gapless decoding is disabled for now, since it does not work well with seeking */
+ mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_SEEKBUFFER, 0); /* Tells mpg123 to use a small read-ahead buffer for better MPEG sync; essential for MP3 radio streams */
+ mpg123_param (mpg123_decoder->handle, MPG123_RESYNC_LIMIT, -1, 0); /* Sets the resync limit to the end of the stream (e.g. don't give up prematurely) */
+
+ /* Open in feed mode (= encoded data is fed manually into the handle). */
+ error = mpg123_open_feed (mpg123_decoder->handle);
+
+ if (G_UNLIKELY (error != MPG123_OK)) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("Error opening mpg123 feed: %s", mpg123_plain_strerror (error)));
+ mpg123_close (mpg123_decoder->handle);
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ return FALSE;
+ }
+
+ GST_DEBUG_OBJECT (dec, "mpg123 decoder started");
+
+ return TRUE;
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_stop (GstAudioDecoder * dec)
+{
+ GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ if (G_LIKELY (mpg123_decoder->handle != NULL)) {
+ mpg123_close (mpg123_decoder->handle);
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ }
+
+ GST_DEBUG_OBJECT (dec, "mpg123 decoder stopped");
+
+ return TRUE;
+}
+
+
+static GstFlowReturn
+gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder,
+ unsigned char const *decoded_bytes, size_t const num_decoded_bytes)
+{
+ GstBuffer *output_buffer;
+ GstFlowReturn alloc_error;
+ GstAudioDecoder *dec;
+
+ output_buffer = NULL;
+ dec = GST_AUDIO_DECODER (mpg123_decoder);
+
+ if ((num_decoded_bytes == 0) || (decoded_bytes == NULL)) {
+ /* This occurs in the first few frames, which do not carry data; once MPG123_AUDIO_DEC_NEW_FORMAT is received, the empty frames stop occurring */
+ GST_TRACE_OBJECT (mpg123_decoder,
+ "Nothing was decoded -> no output buffer to push");
+ return GST_FLOW_OK;
+ }
+
+ output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
+ alloc_error = (output_buffer == NULL) ? GST_FLOW_ERROR : GST_FLOW_OK;
+
+ if (alloc_error != GST_FLOW_OK) {
+ /* This is necessary to advance playback in time, even when nothing was decoded. */
+ return gst_audio_decoder_finish_frame (dec, NULL, 1);
+ } else {
+ GstMapInfo info;
+
+ if (gst_buffer_map (output_buffer, &info, GST_MAP_WRITE)) {
+ if (info.size != num_decoded_bytes)
+ GST_ERROR_OBJECT (mpg123_decoder,
+ "Mapped memory region has size %u instead of expected size %u",
+ info.size, num_decoded_bytes);
+ else
+ memcpy (info.data, decoded_bytes, num_decoded_bytes);
+
+ gst_buffer_unmap (output_buffer, &info);
+ } else
+ GST_ERROR_OBJECT (mpg123_decoder, "Could not map buffer");
+
+ return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
+ }
+}
+
+
+static GstFlowReturn
+gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
+{
+ GstMpg123AudioDec *mpg123_decoder;
+ int decode_error;
+ unsigned char *decoded_bytes;
+ size_t num_decoded_bytes;
+
+ if (G_UNLIKELY (!buffer))
+ return GST_FLOW_OK;
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ if (G_UNLIKELY (mpg123_decoder->handle == NULL)) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("mpg123 handle is NULL"));
+ return GST_FLOW_ERROR;
+ }
+
+ /* The actual decoding */
+ {
+ unsigned char const *inmemory;
+ size_t inmemsize;
+ GstMemory *memory;
+ GstMapInfo info;
+
+ memory = gst_buffer_get_all_memory (buffer);
+ if (memory == NULL)
+ return GST_FLOW_ERROR;
+
+ if (!gst_memory_map (memory, &info, GST_MAP_WRITE)) {
+ gst_memory_unref (memory);
+ return GST_FLOW_ERROR;
+ }
+
+ inmemory = info.data;
+ inmemsize = info.size;
+
+ mpg123_feed (mpg123_decoder->handle, inmemory, inmemsize);
+ decoded_bytes = NULL;
+ num_decoded_bytes = 0;
+ decode_error = mpg123_decode_frame (mpg123_decoder->handle,
+ &mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
+
+ gst_memory_unmap (memory, &info);
+ gst_memory_unref (memory);
+ }
+
+ switch (decode_error) {
+ case MPG123_NEW_FORMAT:
+ /*
+ As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo is not set immediately;
+ instead, the code waits for mpg123 to take note of the new format, and then sets the audioinfo
+ This fixes glitches with mp3s containing several format headers (for example, first half using 44.1kHz, second half 32 kHz)
+ */
+
+ GST_DEBUG_OBJECT (dec,
+ "mpg123 reported a new format -> setting next srccaps");
+
+ gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
+ num_decoded_bytes);
+
+ /*
+ If there is a next audioinfo, use it, then set has_next_audioinfo to FALSE, to make sure
+ gst_audio_decoder_set_output_format() isn't called again until set_format is called by the base class
+ */
+ if (mpg123_decoder->has_next_audioinfo) {
+ if (!gst_audio_decoder_set_output_format (dec,
+ &(mpg123_decoder->next_audioinfo))) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("Unable to set output format"));
+ }
+ mpg123_decoder->has_next_audioinfo = FALSE;
+ }
+
+ break;
+
+ case MPG123_NEED_MORE:
+ case MPG123_OK:
+ return gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
+ decoded_bytes, num_decoded_bytes);
+
+ /* If this happens, then the upstream parser somehow missed the ending of the bitstream */
+ case MPG123_DONE:
+ GST_DEBUG_OBJECT (dec, "mpg123 is done decoding");
+ gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
+ num_decoded_bytes);
+ return GST_FLOW_EOS;
+
+ /* Anything else is considered an error */
+ default:
+ {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL), ("Decoding error: %s",
+ mpg123_plain_strerror (decode_error)));
+
+ return GST_FLOW_ERROR;
+ }
+ }
+
+ return GST_FLOW_OK;
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * incoming_caps)
+{
+/*
+ Using the parsed information upstream, and the list of allowed caps downstream, this code
+ tries to find a suitable audio info. It is important to keep in mind that the rate and number of channels
+ should never deviate from the one the bitstream has, otherwise mpg123 has to mix channels and/or
+ resample (and as its docs say, its internal resampler is very crude). The sample format, however,
+ can be chosen freely, because the MPEG specs do not mandate any special format.
+ Therefore, rate and number of channels are taken from upstream (which parsed the MPEG frames, so
+ the incoming_caps contain exactly the rate and number of channels the bitstream actually has), while
+ the sample format is chosen by trying out all caps that are allowed by downstream. This way, the output
+ is adjusted to what the downstream prefers.
+
+ Also, the new output audio info is not set immediately. Instead, it is considered the "next audioinfo".
+ The code waits for mpg123 to notice the new format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
+ and then sets the next audioinfo. Otherwise, the next audioinfo is set too soon, which may cause problems with
+ mp3s containing several format headers. One example would be an mp3 with the first 30 seconds using 44.1 kHz,
+ then the next 30 seconds using 32 kHz. Rare, but possible.
+
+ STEPS:
+
+ 1. get rate and channels from incoming_caps
+ 2. get allowed caps from src pad
+ 3. for each structure in allowed caps:
+ 3.1. take format
+ 3.2. if the combination of format with rate and channels is unsupported by mpg123, go to (3),
+ or exit with error if there are no more structures to try
+ 3.3. create next audioinfo out of rate,channels,format, and exit
+*/
+
+
+ int rate, channels;
+ GstMpg123AudioDec *mpg123_decoder;
+ GstCaps *allowed_srccaps;
+ guint structure_nr;
+ gboolean match_found = FALSE;
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ if (G_UNLIKELY (mpg123_decoder->handle == NULL)) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("mpg123 handle is NULL"));
+ return FALSE;
+ }
+
+ mpg123_decoder->has_next_audioinfo = FALSE;
+
+ /* Get rate and channels from incoming_caps */
+ {
+ GstStructure *structure;
+ gboolean err = FALSE;
+
+ /* Only the first structure is used (multiple incoming structures don't make sense */
+ structure = gst_caps_get_structure (incoming_caps, 0);
+
+ if (!gst_structure_get_int (structure, "rate", &rate)) {
+ err = TRUE;
+ GST_ERROR_OBJECT (dec, "Incoming caps do not have a rate value");
+ }
+ if (!gst_structure_get_int (structure, "channels", &channels)) {
+ err = TRUE;
+ GST_ERROR_OBJECT (dec, "Incoming caps do not have a channel value");
+ }
+
+ if (err)
+ return FALSE;
+ }
+
+ /* Get the caps that are allowed by downstream */
+ {
+ GstCaps *allowed_srccaps_unnorm =
+ gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
+ allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
+ /* TODO: this causes errors with 1.0 - perhaps a bug? */
+ /*gst_caps_unref(allowed_srccaps_unnorm); */
+ }
+
+ /* Go through all allowed caps, pick the first one that matches */
+ for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
+ ++structure_nr) {
+ GstStructure *structure;
+ gchar const *format_str;
+ GstAudioFormat format;
+ int encoding;
+
+ structure = gst_caps_get_structure (allowed_srccaps, structure_nr);
+
+ format_str = gst_structure_get_string (structure, "format");
+ if (format_str == NULL) {
+ GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
+ continue;
+ }
+
+ format = gst_audio_format_from_string (format_str);
+ if (format == GST_AUDIO_FORMAT_UNKNOWN) {
+ GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
+ continue;
+ }
+
+ switch (format) {
+#ifdef MPG123_ENC_SIGNED_16
+ case GST_AUDIO_FORMAT_S16:
+ encoding = MPG123_ENC_SIGNED_16;
+ break;
+#endif
+#ifdef MPG123_ENC_SIGNED_24
+ case GST_AUDIO_FORMAT_S24:
+ encoding = MPG123_ENC_SIGNED_24;
+ break;
+#endif
+#ifdef MPG123_ENC_SIGNED_32
+ case GST_AUDIO_FORMAT_S32:
+ encoding = MPG123_ENC_SIGNED_32;
+ break;
+#endif
+#ifdef MPG123_ENC_UNSIGNED_16
+ case GST_AUDIO_FORMAT_U16:
+ encoding = MPG123_ENC_UNSIGNED_16;
+ break;
+#endif
+#ifdef MPG123_ENC_UNSIGNED_24
+ case GST_AUDIO_FORMAT_U24:
+ encoding = MPG123_ENC_UNSIGNED_24;
+ break;
+#endif
+#ifdef MPG123_ENC_UNSIGNED_32
+ case GST_AUDIO_FORMAT_U32:
+ encoding = MPG123_ENC_UNSIGNED_32;
+ break;
+#endif
+#ifdef MPG123_ENC_FLOAT_32
+ case GST_AUDIO_FORMAT_F32:
+ encoding = MPG123_ENC_FLOAT_32;
+ break;
+#endif
+ default:
+ GST_DEBUG_OBJECT (dec,
+ "Format %s in srccaps is not supported by mpg123", format_str);
+ continue;
+ }
+
+ {
+ int err;
+
+ /* Cleanup old formats & set new one */
+ mpg123_format_none (mpg123_decoder->handle);
+ err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
+ if (err != MPG123_OK) {
+ GST_DEBUG_OBJECT (dec,
+ "mpg123 cannot use caps %" GST_PTR_FORMAT
+ " because mpg123_format() failed: %s", structure,
+ mpg123_plain_strerror (err));
+ continue;
+ }
+ }
+
+ gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
+ gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
+ channels, NULL);
+ GST_DEBUG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
+ format_str, rate, channels);
+ mpg123_decoder->has_next_audioinfo = TRUE;
+
+ match_found = TRUE;
+
+ break;
+ }
+
+ gst_caps_unref (allowed_srccaps);
+
+ return match_found;
+}
+
+
+static void
+gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard)
+{
+ int error;
+ GstMpg123AudioDec *mpg123_decoder;
+
+ hard = hard;
+
+ GST_DEBUG_OBJECT (dec, "Flushing decoder");
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ if (G_UNLIKELY (mpg123_decoder->handle == NULL)) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("mpg123 handle is NULL"));
+ return;
+ }
+
+ /* Flush by reopening the feed */
+ mpg123_close (mpg123_decoder->handle);
+ error = mpg123_open_feed (mpg123_decoder->handle);
+
+ if (G_UNLIKELY (error != MPG123_OK)) {
+ GstElement *element = GST_ELEMENT (dec);
+ GST_ELEMENT_ERROR (element, STREAM, DECODE, (NULL),
+ ("Error reopening mpg123 feed: %s", mpg123_plain_strerror (error)));
+ mpg123_close (mpg123_decoder->handle);
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ }
+
+ mpg123_decoder->has_next_audioinfo = FALSE;
+
+ /*
+ opening/closing feeds do not affect the format defined by the mpg123_format() call that was made in gst_mpg123_audio_dec_set_format(),
+ and since the up/downstream caps are not expected to change here, no mpg123_format() calls are done
+ */
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "mpg123audiodec",
+ GST_RANK_NONE, gst_mpg123_audio_dec_get_type ());
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ mpg123, "mp3 decoding based on the mpg123 library",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
projects / platform / upstream / gstreamer.git / commitdiff