diff options
| -rw-r--r-- | ext/mpg123/Makefile.am | 11 | ||||
| -rw-r--r-- | ext/mpg123/gstmpg123audiodec.c | 634 | ||||
| -rw-r--r-- | ext/mpg123/gstmpg123audiodec.h | 74 | ||||
| -rw-r--r-- | ext/mpg123/meson.build | 16 | ||||
| -rw-r--r-- | tests/check/elements/mpg123audiodec.c | 534 |
5 files changed, 1269 insertions, 0 deletions
diff --git a/ext/mpg123/Makefile.am b/ext/mpg123/Makefile.am new file mode 100644 index 000000000..465f32597 --- /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 000000000..fa6743cb9 --- /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 000000000..b865c417a --- /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 000000000..a575449bd --- /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 000000000..20d6e779d --- /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) |