Moving mpg123 plugin from -ugly
diff --git a/ext/mpg123/Makefile.am b/ext/mpg123/Makefile.am
new file mode 100644
index 0000000..465f325
--- /dev/null
+++ b/ext/mpg123/Makefile.am
@@ -0,0 +1,11 @@
+plugin_LTLIBRARIES = libgstmpg123.la
+
+libgstmpg123_la_SOURCES = gstmpg123audiodec.c
+libgstmpg123_la_CFLAGS = -DGST_USE_UNSTABLE_API \
+ $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(MPG123_CFLAGS)
+libgstmpg123_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_API_VERSION@ \
+ $(GST_BASE_LIBS) $(GST_LIBS) $(MPG123_LIBS)
+libgstmpg123_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstmpg123audiodec.h
diff --git a/ext/mpg123/gstmpg123audiodec.c b/ext/mpg123/gstmpg123audiodec.c
new file mode 100644
index 0000000..fa6743c
--- /dev/null
+++ b/ext/mpg123/gstmpg123audiodec.c
@@ -0,0 +1,634 @@
+/* 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
+ */
+
+/**
+ * SECTION: element-mpg123audiodec
+ * @see_also: lamemp3enc, mad
+ *
+ * Audio decoder for MPEG-1 layer 1/2/3 audio data.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Decode and play the mp3 file
+ * </refsect2>
+ */
+
+#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 static_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 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 * input_buffer);
+static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec,
+ GstCaps * input_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)
+{
+ GstAudioDecoderClass *base_class;
+ GstElementClass *element_class;
+ GstPadTemplate *src_template, *sink_template;
+ int error;
+
+ GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
+
+ base_class = GST_AUDIO_DECODER_CLASS (klass);
+ element_class = GST_ELEMENT_CLASS (klass);
+
+ 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 */
+ {
+ const int *format_list;
+ const long *rates_list;
+ size_t num, i;
+ GString *s;
+ GstCaps *src_template_caps;
+
+ s = g_string_new ("audio/x-raw, ");
+
+ mpg123_encodings (&format_list, &num);
+ g_string_append (s, "format = { ");
+ for (i = 0; i < num; ++i) {
+ switch (format_list[i]) {
+ case MPG123_ENC_SIGNED_16:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (S16));
+ break;
+ case MPG123_ENC_UNSIGNED_16:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (U16));
+ break;
+ case MPG123_ENC_SIGNED_24:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (S24));
+ break;
+ case MPG123_ENC_UNSIGNED_24:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (U24));
+ break;
+ case MPG123_ENC_SIGNED_32:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (S32));
+ break;
+ case MPG123_ENC_UNSIGNED_32:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (U32));
+ break;
+ case MPG123_ENC_FLOAT_32:
+ g_string_append (s, (i > 0) ? ", " : "");
+ g_string_append (s, GST_AUDIO_NE (F32));
+ break;
+ default:
+ GST_DEBUG ("Ignoring mpg123 format %d", format_list[i]);
+ break;
+ }
+ }
+ g_string_append (s, " }, ");
+
+ mpg123_rates (&rates_list, &num);
+ g_string_append (s, "rate = (int) { ");
+ for (i = 0; i < num; ++i) {
+ g_string_append_printf (s, "%s%lu", (i > 0) ? ", " : "", rates_list[i]);
+ }
+ g_string_append (s, "}, ");
+
+ g_string_append (s, "channels = (int) [ 1, 2 ], ");
+ g_string_append (s, "layout = (string) interleaved");
+
+ src_template_caps = gst_caps_from_string (s->str);
+ src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+ src_template_caps);
+ gst_caps_unref (src_template_caps);
+
+ g_string_free (s, TRUE);
+ }
+
+ sink_template = gst_static_pad_template_get (&static_sink_template);
+
+ gst_element_class_add_pad_template (element_class, 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_INFO ("mpg123 library initialized");
+}
+
+
+void
+gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
+{
+ mpg123_decoder->handle = NULL;
+ gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE);
+ gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
+ (mpg123_decoder), TRUE);
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (mpg123_decoder));
+}
+
+
+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);
+
+ /* 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_REMOVE_FLAGS, MPG123_GAPLESS, 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_ADD_FLAGS, MPG123_SEEKBUFFER, 0);
+ /* Sets the resync limit to the end of the stream (otherwise mpg123 may give
+ * up on decoding prematurely, especially with mp3 web radios) */
+ mpg123_param (mpg123_decoder->handle, MPG123_RESYNC_LIMIT, -1, 0);
+#if MPG123_API_VERSION >= 36
+ /* The precise API version where MPG123_AUTO_RESAMPLE appeared is
+ * somewhere between 29 and 36 */
+ /* Don't let mpg123 resample output */
+ mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS,
+ MPG123_AUTO_RESAMPLE, 0);
+#endif
+ /* Don't let mpg123 print messages to stdout/stderr */
+ mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0);
+
+ /* 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)) {
+ GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
+ ("%s", mpg123_strerror (mpg123_decoder->handle)));
+ mpg123_close (mpg123_decoder->handle);
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ return FALSE;
+ }
+
+ GST_INFO_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_INFO_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;
+ 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_DEBUG_OBJECT (mpg123_decoder,
+ "cannot decode yet, need more data -> no output buffer to push");
+ return GST_FLOW_OK;
+ }
+
+ output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
+
+ if (output_buffer == NULL) {
+ /* 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)) {
+ memcpy (info.data, decoded_bytes, num_decoded_bytes);
+ gst_buffer_unmap (output_buffer, &info);
+ } else {
+ GST_ERROR_OBJECT (mpg123_decoder, "gst_buffer_map() returned NULL");
+ gst_buffer_unref (output_buffer);
+ output_buffer = NULL;
+ }
+
+ return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
+ }
+}
+
+
+static GstFlowReturn
+gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * input_buffer)
+{
+ GstMpg123AudioDec *mpg123_decoder;
+ int decode_error;
+ unsigned char *decoded_bytes;
+ size_t num_decoded_bytes;
+ GstFlowReturn retval;
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ g_assert (mpg123_decoder->handle != NULL);
+
+ /* The actual decoding */
+ {
+ /* feed input data (if there is any) */
+ if (G_LIKELY (input_buffer != NULL)) {
+ GstMapInfo info;
+
+ if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
+ mpg123_feed (mpg123_decoder->handle, info.data, info.size);
+ gst_buffer_unmap (input_buffer, &info);
+ } else {
+ GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, RESOURCE, READ, (NULL),
+ ("gst_memory_map() failed"), retval);
+ return retval;
+ }
+ }
+
+ /* Try to decode a frame */
+ decoded_bytes = NULL;
+ num_decoded_bytes = 0;
+ decode_error = mpg123_decode_frame (mpg123_decoder->handle,
+ &mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
+ }
+
+ retval = GST_FLOW_OK;
+
+ 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_LOG_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))) {
+ GST_WARNING_OBJECT (dec, "Unable to set output format");
+ retval = GST_FLOW_NOT_NEGOTIATED;
+ }
+ mpg123_decoder->has_next_audioinfo = FALSE;
+ }
+
+ break;
+
+ case MPG123_NEED_MORE:
+ case MPG123_OK:
+ retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
+ decoded_bytes, num_decoded_bytes);
+ break;
+
+ case MPG123_DONE:
+ /* If this happens, then the upstream parser somehow missed the ending
+ * of the bitstream */
+ GST_LOG_OBJECT (dec, "mpg123 is done decoding");
+ gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
+ num_decoded_bytes);
+ retval = GST_FLOW_EOS;
+ break;
+
+ default:
+ {
+ /* Anything else is considered an error */
+ int errcode;
+ retval = GST_FLOW_ERROR; /* use error by default */
+ switch (decode_error) {
+ case MPG123_ERR:
+ errcode = mpg123_errcode (mpg123_decoder->handle);
+ break;
+ default:
+ errcode = decode_error;
+ }
+ switch (errcode) {
+ case MPG123_BAD_OUTFORMAT:{
+ GstCaps *input_caps =
+ gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec));
+ GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
+ ("Output sample format could not be used when trying to decode frame. "
+ "This is typically caused when the input caps (often the sample "
+ "rate) do not match the actual format of the audio data. "
+ "Input caps: %" GST_PTR_FORMAT, input_caps)
+ );
+ gst_caps_unref (input_caps);
+ break;
+ }
+ default:{
+ char const *errmsg = mpg123_plain_strerror (errcode);
+ /* GST_AUDIO_DECODER_ERROR sets a new return value according to
+ * its estimations */
+ GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, STREAM, DECODE, (NULL),
+ ("mpg123 decoding error: %s", errmsg), retval);
+ }
+ }
+ }
+ }
+
+ return retval;
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
+{
+ /* "encoding" is the sample format specifier for mpg123 */
+ int encoding;
+ int sample_rate, num_channels;
+ GstAudioFormat format;
+ GstMpg123AudioDec *mpg123_decoder;
+ gboolean retval = FALSE;
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ g_assert (mpg123_decoder->handle != NULL);
+
+ mpg123_decoder->has_next_audioinfo = FALSE;
+
+ /* Get sample rate and number of channels from input_caps */
+ {
+ GstStructure *structure;
+ gboolean err = FALSE;
+
+ /* Only the first structure is used (multiple
+ * input caps structures don't make sense */
+ structure = gst_caps_get_structure (input_caps, 0);
+
+ if (!gst_structure_get_int (structure, "rate", &sample_rate)) {
+ err = TRUE;
+ GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
+ }
+ if (!gst_structure_get_int (structure, "channels", &num_channels)) {
+ err = TRUE;
+ GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
+ }
+
+ if (G_UNLIKELY (err))
+ goto done;
+ }
+
+ /* Get sample format from the allowed src caps */
+ {
+ GstCaps *allowed_srccaps =
+ gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
+
+ if (allowed_srccaps == NULL) {
+ /* srcpad is not linked (yet), so no peer information is available;
+ * just use the default sample format (16 bit signed integer) */
+ GST_DEBUG_OBJECT (mpg123_decoder,
+ "srcpad is not linked (yet) -> using S16 sample format");
+ format = GST_AUDIO_FORMAT_S16;
+ encoding = MPG123_ENC_SIGNED_16;
+ } else if (gst_caps_is_empty (allowed_srccaps)) {
+ gst_caps_unref (allowed_srccaps);
+ goto done;
+ } else {
+ gchar const *format_str;
+ GValue const *format_value;
+
+ /* Look at the sample format values from the first structure */
+ GstStructure *structure = gst_caps_get_structure (allowed_srccaps, 0);
+ format_value = gst_structure_get_value (structure, "format");
+
+ if (format_value == NULL) {
+ gst_caps_unref (allowed_srccaps);
+ goto done;
+ } else if (GST_VALUE_HOLDS_LIST (format_value)) {
+ /* if value is a format list, pick the first entry */
+ GValue const *fmt_list_value =
+ gst_value_list_get_value (format_value, 0);
+ format_str = g_value_get_string (fmt_list_value);
+ } else if (G_VALUE_HOLDS_STRING (format_value)) {
+ /* if value is a string, use it directly */
+ format_str = g_value_get_string (format_value);
+ } else {
+ GST_ERROR_OBJECT (mpg123_decoder, "unexpected type for 'format' field "
+ "in caps structure %" GST_PTR_FORMAT, structure);
+ gst_caps_unref (allowed_srccaps);
+ goto done;
+ }
+
+ /* get the format value from the string */
+ format = gst_audio_format_from_string (format_str);
+ gst_caps_unref (allowed_srccaps);
+
+ g_assert (format != GST_AUDIO_FORMAT_UNKNOWN);
+
+ /* convert format to mpg123 encoding */
+ switch (format) {
+ case GST_AUDIO_FORMAT_S16:
+ encoding = MPG123_ENC_SIGNED_16;
+ break;
+ case GST_AUDIO_FORMAT_S24:
+ encoding = MPG123_ENC_SIGNED_24;
+ break;
+ case GST_AUDIO_FORMAT_S32:
+ encoding = MPG123_ENC_SIGNED_32;
+ break;
+ case GST_AUDIO_FORMAT_U16:
+ encoding = MPG123_ENC_UNSIGNED_16;
+ break;
+ case GST_AUDIO_FORMAT_U24:
+ encoding = MPG123_ENC_UNSIGNED_24;
+ break;
+ case GST_AUDIO_FORMAT_U32:
+ encoding = MPG123_ENC_UNSIGNED_32;
+ break;
+ case GST_AUDIO_FORMAT_F32:
+ encoding = MPG123_ENC_FLOAT_32;
+ break;
+ default:
+ g_assert_not_reached ();
+ goto done;
+ }
+ }
+ }
+
+ /* Sample rate, number of channels, and sample format are known at this point.
+ * Set the audioinfo structure's values and the mpg123 format. */
+ {
+ int err;
+
+ /* clear all existing format settings from the mpg123 instance */
+ mpg123_format_none (mpg123_decoder->handle);
+ /* set the chosen format */
+ err =
+ mpg123_format (mpg123_decoder->handle, sample_rate, num_channels,
+ encoding);
+
+ if (err != MPG123_OK) {
+ GST_WARNING_OBJECT (dec,
+ "mpg123_format() failed: %s",
+ mpg123_strerror (mpg123_decoder->handle));
+ } else {
+ gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
+ gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format,
+ sample_rate, num_channels, NULL);
+ GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
+ gst_audio_format_to_string (format), sample_rate, num_channels);
+ mpg123_decoder->has_next_audioinfo = TRUE;
+
+ retval = TRUE;
+ }
+ }
+
+done:
+ return retval;
+}
+
+
+static void
+gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard)
+{
+ int error;
+ GstMpg123AudioDec *mpg123_decoder;
+
+ GST_LOG_OBJECT (dec, "Flushing decoder");
+
+ mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+ g_assert (mpg123_decoder->handle != NULL);
+
+ /* Flush by reopening the feed */
+ mpg123_close (mpg123_decoder->handle);
+ error = mpg123_open_feed (mpg123_decoder->handle);
+
+ if (G_UNLIKELY (error != MPG123_OK)) {
+ GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
+ ("Error while reopening mpg123 feed: %s",
+ mpg123_plain_strerror (error)));
+ mpg123_close (mpg123_decoder->handle);
+ mpg123_delete (mpg123_decoder->handle);
+ mpg123_decoder->handle = NULL;
+ }
+
+ if (hard)
+ 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_MARGINAL, 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)
diff --git a/ext/mpg123/gstmpg123audiodec.h b/ext/mpg123/gstmpg123audiodec.h
new file mode 100644
index 0000000..b865c41
--- /dev/null
+++ b/ext/mpg123/gstmpg123audiodec.h
@@ -0,0 +1,74 @@
+/* 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
+ */
+
+#ifndef __GST_MPG123_AUDIO_DEC_H__
+#define __GST_MPG123_AUDIO_DEC_H__
+
+/* This is what the visual studio build in mpg123 does before including the
+ * original header file. Without this we get syntax errors in the
+ * replace_reader function declarations because it doesn't know ssize_t etc.
+ * It doesn't realy matter for us if the ssize_t typedef here is correct. */
+#ifdef _MSC_VER
+#include <tchar.h>
+#include <stdlib.h>
+#include <sys/types.h>
+typedef long ssize_t;
+#include <stdint.h>
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
+#include <mpg123.h>
+
+
+G_BEGIN_DECLS
+
+
+typedef struct _GstMpg123AudioDec GstMpg123AudioDec;
+typedef struct _GstMpg123AudioDecClass GstMpg123AudioDecClass;
+
+
+#define GST_TYPE_MPG123_AUDIO_DEC (gst_mpg123_audio_dec_get_type())
+#define GST_MPG123_AUDIO_DEC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_MPG123_AUDIO_DEC,GstMpg123AudioDec))
+#define GST_MPG123_AUDIO_DEC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_MPG123_AUDIO_DEC,GstMpg123AudioDecClass))
+#define GST_IS_MPG123_AUDIO_DEC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_MPG123_AUDIO_DEC))
+#define GST_IS_MPG123_AUDIO_DEC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_MPG123_AUDIO_DEC))
+
+struct _GstMpg123AudioDec
+{
+ GstAudioDecoder parent;
+
+ mpg123_handle *handle;
+
+ GstAudioInfo next_audioinfo;
+ gboolean has_next_audioinfo;
+
+ off_t frame_offset;
+};
+
+
+struct _GstMpg123AudioDecClass
+{
+ GstAudioDecoderClass parent_class;
+};
+
+G_GNUC_INTERNAL GType gst_mpg123_audio_dec_get_type (void);
+
+G_END_DECLS
+
+#endif
diff --git a/ext/mpg123/meson.build b/ext/mpg123/meson.build
new file mode 100644
index 0000000..a575449
--- /dev/null
+++ b/ext/mpg123/meson.build
@@ -0,0 +1,16 @@
+mpg123_sources = [
+ 'gstmpg123audiodec.c',
+]
+
+mpg123_dep = dependency('libmpg123', version : '>= 1.3', required : false)
+
+if mpg123_dep.found()
+ gstmpg123 = library('gstmpg123',
+ mpg123_sources,
+ c_args : ugly_args,
+ include_directories : [configinc],
+ dependencies : [gstaudio_dep, mpg123_dep],
+ install : true,
+ install_dir : plugins_install_dir,
+ )
+endif
diff --git a/tests/check/elements/mpg123audiodec.c b/tests/check/elements/mpg123audiodec.c
new file mode 100644
index 0000000..20d6e77
--- /dev/null
+++ b/tests/check/elements/mpg123audiodec.c
@@ -0,0 +1,534 @@
+/* GStreamer
+ *
+ * unit test for mpg123audiodec
+ *
+ * Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.org>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/audio.h>
+
+#include <gst/fft/gstfft.h>
+#include <gst/fft/gstffts16.h>
+#include <gst/fft/gstffts32.h>
+#include <gst/fft/gstfftf32.h>
+#include <gst/fft/gstfftf64.h>
+
+#include <gst/app/gstappsink.h>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+
+#define MP2_STREAM_FILENAME "stream.mp2"
+#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
+#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
+
+
+/* mpeg 1 layer 2 stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ * avenc_mp2 bitrate=32000 ! tee name=t \
+ * t. ! queue ! fakesink silent=false \
+ * t. ! queue ! filesink location=test.mp2
+ *
+ * mpeg 1 layer 3 CBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ * lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
+ * "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ * t. ! queue ! fakesink silent=false \
+ * t. ! queue ! filesink location=test.mp3
+ *
+ * mpeg 1 layer 3 VBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ * lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
+ * "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ * t. ! queue ! fakesink silent=false \
+ * t. ! queue ! filesink location=test.mp3
+ */
+
+
+/* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
+
+#define FFT_HELPERS(type,ffttag,ffttag2,scale) \
+static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
+{ \
+ gdouble mag = (gdouble) c->r * (gdouble) c->r; \
+ mag += (gdouble) c->i * (gdouble) c->i; \
+ mag /= scale * scale; \
+ mag = 10.0 * log10 (mag); \
+ return mag; \
+} \
+static gdouble find_main_frequency_spot_##ffttag ( \
+ const GstFFT##ffttag##Complex *v, int elements) \
+{ \
+ int i; \
+ gdouble maxmag = -9999; \
+ int maxidx = 0; \
+ for (i=0; i<elements; ++i) { \
+ gdouble mag = magnitude##ffttag (v+i); \
+ if (mag > maxmag) { \
+ maxmag = mag; \
+ maxidx = i; \
+ } \
+ } \
+ return maxidx / (gdouble) elements; \
+} \
+static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, \
+ int elements, gdouble spot) \
+{ \
+ int i; \
+ for (i=0; i<elements; ++i) { \
+ gdouble pos = i / (gdouble) elements; \
+ gdouble mag = magnitude##ffttag (v+i); \
+ if (fabs (pos - spot) > 0.01) { \
+ if (mag > -35.0) { \
+ GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
+ return FALSE; \
+ } \
+ } \
+ } \
+ return TRUE; \
+} \
+static void check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble \
+ expected_spot) \
+{ \
+ GstMapInfo map; \
+ int num_samples; \
+ gdouble actual_spot; \
+ GstFFT##ffttag *ctx; \
+ GstFFT##ffttag##Complex *fftdata; \
+ \
+ gst_buffer_map (buffer, &map, GST_MAP_READ); \
+ \
+ num_samples = map.size / sizeof(type) & ~1; \
+ ctx = gst_fft_##ffttag2##_new (num_samples, FALSE); \
+ fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1); \
+ \
+ gst_fft_##ffttag2##_window (ctx, (type*)map.data, \
+ GST_FFT_WINDOW_HAMMING); \
+ gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata); \
+ \
+ actual_spot = find_main_frequency_spot_##ffttag (fftdata, \
+ num_samples / 2 + 1); \
+ GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
+ fabs (expected_spot - actual_spot)); \
+ fail_unless (fabs (expected_spot - actual_spot) < 0.05, \
+ "Actual main frequency spot is too far away from expected one"); \
+ fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1, \
+ actual_spot), "One secondary peak in spectrum exceeds threshold"); \
+ \
+ gst_buffer_unmap (buffer, &map); \
+ \
+ gst_fft_##ffttag2##_free (ctx); \
+ g_free (fftdata); \
+}
+FFT_HELPERS (gint32, S32, s32, 2147483647.0);
+
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S32))
+ );
+static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS_ANY);
+static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS_ANY);
+
+
+static void
+setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
+ GstElement ** appsink)
+{
+ GstElement *source, *parser;
+
+ *pipeline = gst_pipeline_new (NULL);
+ source = gst_element_factory_make ("filesrc", NULL);
+ parser = gst_element_factory_make ("mpegaudioparse", NULL);
+ *appsink = gst_element_factory_make ("appsink", NULL);
+
+ gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
+ gst_element_link_many (source, parser, *appsink, NULL);
+
+ {
+ char *full_filename =
+ g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
+ g_object_set (G_OBJECT (source), "location", full_filename, NULL);
+ g_free (full_filename);
+ }
+
+ gst_element_set_state (*pipeline, GST_STATE_PLAYING);
+}
+
+static void
+cleanup_input_pipeline (GstElement * pipeline)
+{
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+}
+
+static GstElement *
+setup_mpeg1layer2dec (void)
+{
+ GstElement *mpg123audiodec;
+ GstCaps *caps;
+
+ GST_DEBUG ("setup_mpeg1layer2dec");
+ mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+ mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
+ mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ /* This is necessary to trigger a set_format call in the decoder;
+ * fixed caps don't trigger it */
+ caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "layer", G_TYPE_INT, 2,
+ "rate", G_TYPE_INT, 44100,
+ "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
+ gst_caps_unref (caps);
+
+ return mpg123audiodec;
+}
+
+static GstElement *
+setup_mpeg1layer3dec (void)
+{
+ GstElement *mpg123audiodec;
+ GstCaps *caps;
+
+ GST_DEBUG ("setup_mpeg1layer3dec");
+ mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+ mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
+ mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ /* This is necessary to trigger a set_format call in the decoder;
+ * fixed caps don't trigger it */
+ caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "layer", G_TYPE_INT, 3,
+ "rate", G_TYPE_INT, 44100,
+ "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
+ gst_caps_unref (caps);
+
+ return mpg123audiodec;
+}
+
+static void
+cleanup_mpg123audiodec (GstElement * mpg123audiodec)
+{
+ GST_DEBUG ("cleanup_mpeg1layer2dec");
+ gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (mpg123audiodec);
+ gst_check_teardown_sink_pad (mpg123audiodec);
+ gst_check_teardown_element (mpg123audiodec);
+}
+
+static void
+run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
+{
+ GstBus *bus;
+ unsigned int num_input_buffers, num_decoded_buffers;
+ gint expected_size;
+ GstCaps *out_caps, *caps;
+ GstAudioInfo audioinfo;
+ GstElement *input_pipeline, *input_appsink;
+ int i;
+ GstBuffer *outbuffer;
+
+ /* 440 Hz = frequency of sine wave in audio data
+ * 44100 Hz = sample rate
+ * (44100 / 2) Hz = Nyquist frequency */
+ static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
+
+ fail_unless (gst_element_set_state (mpg123audiodec,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+ bus = gst_bus_new ();
+
+ gst_element_set_bus (mpg123audiodec, bus);
+
+ setup_input_pipeline (filename, &input_pipeline, &input_appsink);
+
+ num_input_buffers = 0;
+ while (TRUE) {
+ GstSample *sample;
+ GstBuffer *input_buffer;
+
+ sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
+ if (sample == NULL)
+ break;
+
+ fail_unless (GST_IS_SAMPLE (sample));
+
+ input_buffer = gst_sample_get_buffer (sample);
+ fail_if (input_buffer == NULL);
+
+ /* This is done to be on the safe side - docs say lifetime of the input buffer
+ * depends *solely* on the sample */
+ input_buffer = gst_buffer_copy (input_buffer);
+
+ fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
+
+ ++num_input_buffers;
+
+ gst_sample_unref (sample);
+ }
+
+ num_decoded_buffers = g_list_length (buffers);
+
+ /* check number of decoded buffers */
+ fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
+
+ caps = gst_pad_get_current_caps (mysinkpad);
+ GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
+ fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
+ "Getting audio info from caps failed");
+
+ /* check caps */
+ out_caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
+ "layout", G_TYPE_STRING, "interleaved",
+ "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
+
+ fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
+
+ gst_caps_unref (out_caps);
+ gst_caps_unref (caps);
+
+ /* here, test if decoded data is a sine tone, and if the sine frequency is at the
+ * right spot in the spectrum */
+ for (i = 0; i < num_decoded_buffers; ++i) {
+ outbuffer = GST_BUFFER (buffers->data);
+ fail_if (outbuffer == NULL, "Invalid buffer retrieved");
+
+ /* MPEG 1 layer 2 uses 1152 samples per frame */
+ expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
+ fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
+
+ check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
+
+ buffers = g_list_remove (buffers, outbuffer);
+ gst_buffer_unref (outbuffer);
+ outbuffer = NULL;
+ }
+
+ g_list_free (buffers);
+ buffers = NULL;
+
+ cleanup_input_pipeline (input_pipeline);
+ gst_bus_set_flushing (bus, TRUE);
+ gst_element_set_bus (mpg123audiodec, NULL);
+ gst_object_unref (GST_OBJECT (bus));
+}
+
+
+GST_START_TEST (test_decode_mpeg1layer2)
+{
+ GstElement *mpg123audiodec;
+ mpg123audiodec = setup_mpeg1layer2dec ();
+ run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
+ cleanup_mpg123audiodec (mpg123audiodec);
+ mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_cbr)
+{
+ GstElement *mpg123audiodec;
+ mpg123audiodec = setup_mpeg1layer3dec ();
+ run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
+ cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_vbr)
+{
+ GstElement *mpg123audiodec;
+ mpg123audiodec = setup_mpeg1layer3dec ();
+ run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
+ cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer2)
+{
+ GstElement *mpg123audiodec;
+ GstBuffer *inbuffer;
+ GstBus *bus;
+ int i, num_buffers;
+ guint32 *tmpbuf;
+
+ mpg123audiodec = setup_mpeg1layer2dec ();
+
+ fail_unless (gst_element_set_state (mpg123audiodec,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+ bus = gst_bus_new ();
+
+ /* initialize the buffer with something that is no mpeg2 */
+ tmpbuf = g_new (guint32, 4096);
+ for (i = 0; i < 4096; i++) {
+ tmpbuf[i] = i;
+ }
+ inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ gst_element_set_bus (mpg123audiodec, bus);
+
+ /* should be possible to push without problems but nothing gets decoded */
+ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+ num_buffers = g_list_length (buffers);
+
+ /* should be 0 buffers as decoding should've been impossible */
+ fail_unless_equals_int (num_buffers, 0);
+
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_bus_set_flushing (bus, TRUE);
+ gst_element_set_bus (mpg123audiodec, NULL);
+ gst_object_unref (GST_OBJECT (bus));
+ cleanup_mpg123audiodec (mpg123audiodec);
+ mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer3)
+{
+ GstElement *mpg123audiodec;
+ GstBuffer *inbuffer;
+ GstBus *bus;
+ int i, num_buffers;
+ guint32 *tmpbuf;
+
+ mpg123audiodec = setup_mpeg1layer3dec ();
+
+ fail_unless (gst_element_set_state (mpg123audiodec,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+ bus = gst_bus_new ();
+
+ /* initialize the buffer with something that is no mpeg2 */
+ tmpbuf = g_new (guint32, 4096);
+ for (i = 0; i < 4096; i++) {
+ tmpbuf[i] = i;
+ }
+ inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ gst_element_set_bus (mpg123audiodec, bus);
+
+ /* should be possible to push without problems but nothing gets decoded */
+ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+ num_buffers = g_list_length (buffers);
+
+ /* should be 0 buffers as decoding should've been impossible */
+ fail_unless_equals_int (num_buffers, 0);
+
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_bus_set_flushing (bus, TRUE);
+ gst_element_set_bus (mpg123audiodec, NULL);
+ gst_object_unref (GST_OBJECT (bus));
+ cleanup_mpg123audiodec (mpg123audiodec);
+ mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+static gboolean
+is_test_file_available (gchar const *filename)
+{
+ gboolean ret;
+ gchar *full_filename;
+ gchar *cwd;
+
+ cwd = g_get_current_dir ();
+ full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
+ ret =
+ g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
+ g_free (full_filename);
+ g_free (cwd);
+ return ret;
+}
+
+static Suite *
+mpg123audiodec_suite (void)
+{
+ GstRegistry *registry;
+ Suite *s = suite_create ("mpg123audiodec");
+ TCase *tc_chain = tcase_create ("general");
+
+ registry = gst_registry_get ();
+
+ suite_add_tcase (s, tc_chain);
+ if (gst_registry_check_feature_version (registry, "filesrc",
+ GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
+ gst_registry_check_feature_version (registry, "mpegaudioparse",
+ GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
+ gst_registry_check_feature_version (registry, "appsrc",
+ GST_VERSION_MAJOR, GST_VERSION_MINOR, 0)) {
+ if (is_test_file_available (MP2_STREAM_FILENAME))
+ tcase_add_test (tc_chain, test_decode_mpeg1layer2);
+ if (is_test_file_available (MP3_CBR_STREAM_FILENAME))
+ tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
+ if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
+ tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
+ }
+ tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
+ tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
+
+ return s;
+}
+
+
+GST_CHECK_MAIN (mpg123audiodec)
