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Code Issues 1 Projects Releases Wiki Activity

mpg123: move plugin from -bad to -ugly

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=719849
This commit is contained in:
Tim-Philipp Müller 2016-02-16 10:44:33 +00:00
parent 057aca2cda
commit 08d8aefcda
13 changed files with 3 additions and 1322 deletions
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2
Makefile.am
View file

@ -56,6 +56,7 @@ CRUFT_FILES = \
$(top_builddir)/ext/hls/.libs/libgstfragmented* \
$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/ext/jack/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/ext/mpg123/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/audioparsers/.libs/*.{so,dll,DLL,dylib} \
@ -122,6 +123,7 @@ CRUFT_DIRS = \
$(top_srcdir)/ext/ivorbis \
$(top_srcdir)/ext/jack \
$(top_srcdir)/ext/metadata \
$(top_srcdir)/ext/mpg123 \
$(top_srcdir)/ext/swfdec \
$(top_srcdir)/ext/tarkin \
$(top_srcdir)/ext/theora \

10
configure.ac
View file

@ -2455,14 +2455,6 @@ AG_GST_CHECK_FEATURE(MPEG2ENC, [mpeg2enc], mpeg2enc, [
fi
])
dnl *** mpg123 ***
translit(dnm, m, l) AM_CONDITIONAL(USE_MPG123, true)
AG_GST_CHECK_FEATURE(MPG123, [mpg123 audio decoder], mpg123, [
PKG_CHECK_MODULES(MPG123, libmpg123 >= 1.13, HAVE_MPG123="yes", HAVE_MPG123="no")
AC_SUBST(MPG123_CFLAGS)
AC_SUBST(MPG123_LIBS)
])
dnl *** mplex ***
translit(dnm, m, l) AM_CONDITIONAL(USE_MPLEX, true)
AG_GST_CHECK_FEATURE(MPLEX, [mplex], mplex, [
@ -3301,7 +3293,6 @@ AM_CONDITIONAL(USE_LINSYS, false)
AM_CONDITIONAL(USE_MODPLUG, false)
AM_CONDITIONAL(USE_MIMIC, false)
AM_CONDITIONAL(USE_MPEG2ENC, false)
AM_CONDITIONAL(USE_MPG123, false)
AM_CONDITIONAL(USE_MPLEX, false)
AM_CONDITIONAL(USE_MUSEPACK, false)
AM_CONDITIONAL(USE_NAS, false)
@ -3615,7 +3606,6 @@ ext/libvisual/Makefile
ext/Makefile
ext/modplug/Makefile
ext/mpeg2enc/Makefile
ext/mpg123/Makefile
ext/mimic/Makefile
ext/mplex/Makefile
ext/musepack/Makefile

1
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -68,7 +68,6 @@
<xi:include href="xml/element-mpegpsmux.xml" />
<xi:include href="xml/element-mpegtsmux.xml" />
<xi:include href="xml/element-mplex.xml" />
<xi:include href="xml/element-mpg123audiodec.xml" />
<xi:include href="xml/element-neonhttpsrc.xml" />
<xi:include href="xml/element-ofa.xml" />
<xi:include href="xml/element-openalsrc.xml" />

14
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -974,20 +974,6 @@ GST_MPLEX_WAIT
gst_mplex_get_type
</SECTION>
<SECTION>
<FILE>element-mpg123audiodec</FILE>
<TITLE>mpg123audiodec</TITLE>
GstMpg123AudioDec
<SUBSECTION Standard>
GstMpg123AudioDecClass
GST_MPG123_AUDIO_DEC
GST_MPG123_AUDIO_DEC_CLASS
GST_IS_MPG123_AUDIO_DEC
GST_IS_MPG123_AUDIO_DEC_CLASS
GST_TYPE_MPG123_AUDIO_DEC
gst_mpg123_audio_dec_get_type
</SECTION>
<SECTION>
<FILE>element-mxfdemux</FILE>
<TITLE>mxfdemux</TITLE>

34
docs/plugins/inspect/plugin-mpg123.xml
View file

@ -1,34 +0,0 @@
<plugin>
<name>mpg123</name>
<description>mp3 decoding based on the mpg123 library</description>
<filename>../../ext/mpg123/.libs/libgstmpg123.so</filename>
<basename>libgstmpg123.so</basename>
<version>1.7.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>mpg123audiodec</name>
<longname>mpg123 mp3 decoder</longname>
<class>Codec/Decoder/Audio</class>
<description>Decodes mp3 streams using the mpg123 library</description>
<author>Carlos Rafael Giani &lt;dv@pseudoterminal.org&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>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</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string){ S16LE, U16LE, S32LE, U32LE, S24LE, U24LE, F32LE }, rate=(int){ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, channels=(int)[ 1, 2 ], layout=(string)interleaved</details>
</caps>
</pads>
</element>
</elements>
</plugin>

8
ext/Makefile.am
View file

@ -184,12 +184,6 @@ else
MPEG2ENC_DIR=
endif
if USE_MPG123
MPG123_DIR=mpg123
else
MPG123_DIR=
endif
if USE_MIMIC
MIMIC_DIR=mimic
else
@ -446,7 +440,6 @@ SUBDIRS=\
$(LIBVISUAL_DIR) \
$(MODPLUG_DIR) \
$(MPEG2ENC_DIR) \
$(MPG123_DIR) \
$(MIMIC_DIR) \
$(MPLEX_DIR) \
$(MUSEPACK_DIR) \
@ -514,7 +507,6 @@ DIST_SUBDIRS = \
modplug \
mimic \
mpeg2enc \
mpg123 \
mplex \
musepack \
nas \

12
ext/mpg123/Makefile.am
View file

@ -1,12 +0,0 @@
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)
libgstmpg123_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstmpg123audiodec.h

633
ext/mpg123/gstmpg123audiodec.c
View file

@ -1,633 +0,0 @@
/* 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);
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)

62
ext/mpg123/gstmpg123audiodec.h
View file

@ -1,62 +0,0 @@
/* 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__
#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

1
gst-plugins-bad.spec.in
View file

@ -223,7 +223,6 @@ make ERROR_CFLAGS='' ERROR_CXXFLAGS=''
%{_libdir}/gstreamer-%{majorminor}/libgstfreeverb.so
%{_libdir}/gstreamer-%{majorminor}/libgstivtc.so
%{_libdir}/gstreamer-%{majorminor}/libgstmidi.so
%{_libdir}/gstreamer-%{majorminor}/libgstmpg123.so
%{_libdir}/gstreamer-%{majorminor}/libgstrfbsrc.so
%{_libdir}/gstreamer-%{majorminor}/libgstsmoothstreaming.so
%{_libdir}/gstreamer-%{majorminor}/libgstvideofiltersbad.so

12
tests/check/Makefile.am
View file

@ -84,12 +84,6 @@ else
check_mpeg2enc =
endif
if USE_MPG123
check_mpg123 = elements/mpg123audiodec
else
check_mpg123 =
endif
if USE_MPLEX
check_mplex = elements/mplex
else
@ -281,7 +275,6 @@ check_PROGRAMS = \
elements/mpegtsmux \
elements/mpegvideoparse \
elements/mpeg4videoparse \
$(check_mpg123) \
elements/mxfdemux \
elements/mxfmux \
elements/netsim \
@ -455,11 +448,6 @@ elements_assrender_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION
elements_mpegtsmux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_mpegtsmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(GST_BASE_LIBS) $(LDADD)
elements_mpg123audiodec_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_mpg123audiodec_LDADD = \
$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
-lgstaudio-@GST_API_VERSION@ -lgstfft-@GST_API_VERSION@ -lgstapp-@GST_API_VERSION@
elements_uvch264demux_CFLAGS = -DUVCH264DEMUX_DATADIR="$(srcdir)/elements/uvch264demux_data" \
$(AM_CFLAGS)

2
tests/check/elements/.gitignore vendored
View file

@ -38,12 +38,12 @@ mpeg2enc
mpegvideoparse
mpeg4videoparse
mpegtsmux
mpg123audiodec
mplex
mssdemux
mxfdemux
mxfmux
neonhttpsrc
netsim
ofa
opus
pcapparse

534
tests/check/elements/mpg123audiodec.c
View file

@ -1,534 +0,0 @@
/* 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)