Remove mpg123 plugin, moved to -good

https://bugzilla.gnome.org/show_bug.cgi?id=774252
This commit is contained in:
Tim-Philipp Müller 2017-08-20 14:31:02 +01:00
parent 53160e8fa1
commit 83ff57c849
17 changed files with 3 additions and 1359 deletions

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@ -40,10 +40,12 @@ CRUFT_FILES = \
$(top_builddir)/gst-plugins-ugly.spec \ $(top_builddir)/gst-plugins-ugly.spec \
$(top_builddir)/common/shave \ $(top_builddir)/common/shave \
$(top_builddir)/common/shave-libtool \ $(top_builddir)/common/shave-libtool \
$(top_builddir)/ext/mpg123/.libs/libgstmpg123.so \
$(top_builddir)/gst/realmedia/.libs/libgstrmdemux.so $(top_builddir)/gst/realmedia/.libs/libgstrmdemux.so
CRUFT_DIRS = \ CRUFT_DIRS = \
$(top_srcdir)/docs/plugins/tmpl \ $(top_srcdir)/docs/plugins/tmpl \
$(top_srcdir)/ext/mpg123/ \
$(top_builddir)/win32 \ $(top_builddir)/win32 \
$(top_srcdir)/win32 $(top_srcdir)/win32

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@ -9,7 +9,7 @@ Required tools:
=============== ===============
An extra set of tools is required if you wish to build GStreamer out of An extra set of tools is required if you wish to build GStreamer out of
CVS (using autogen.sh): git (using autogen.sh):
autoconf 2.52 or better autoconf 2.52 or better
automake 1.5 automake 1.5
@ -34,8 +34,6 @@ a52dec (for the a52dec AC-3 decoder)
http://liba52.sourceforge.net/ http://liba52.sourceforge.net/
opencore-amr (for the AMR-NB decoder and encoder and the AMR-WB decoder) opencore-amr (for the AMR-NB decoder and encoder and the AMR-WB decoder)
http://sourceforge.net/projects/opencore-amr/ http://sourceforge.net/projects/opencore-amr/
libmpg123 (for the mpg123 mp3 decoder plugin)
https://www.mpg123.de/api/
liblame (for lame mp3 encoder) liblame (for lame mp3 encoder)
http://www.mp3dev.org/mp3/ http://www.mp3dev.org/mp3/
libdvdread (for the dvdreadsrc) libdvdread (for the dvdreadsrc)

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@ -303,14 +303,6 @@ AG_GST_CHECK_FEATURE(MPEG2DEC, [mpeg2dec], mpeg2dec, [
AG_GST_PKG_CHECK_MODULES(MPEG2DEC, libmpeg2 >= 0.5.1) AG_GST_PKG_CHECK_MODULES(MPEG2DEC, libmpeg2 >= 0.5.1)
]) ])
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 *** sidplay : works with libsidplay 1.36.x (not 2.x.x) *** dnl *** sidplay : works with libsidplay 1.36.x (not 2.x.x) ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SIDPLAY, true) translit(dnm, m, l) AM_CONDITIONAL(USE_SIDPLAY, true)
AG_GST_CHECK_FEATURE(SIDPLAY, [libsidplay], sid, [ AG_GST_CHECK_FEATURE(SIDPLAY, [libsidplay], sid, [
@ -357,7 +349,6 @@ AM_CONDITIONAL(USE_CDIO, false)
AM_CONDITIONAL(USE_DVDREAD, false) AM_CONDITIONAL(USE_DVDREAD, false)
AM_CONDITIONAL(USE_LAME, false) AM_CONDITIONAL(USE_LAME, false)
AM_CONDITIONAL(USE_MPEG2DEC, false) AM_CONDITIONAL(USE_MPEG2DEC, false)
AM_CONDITIONAL(USE_MPG123, false)
AM_CONDITIONAL(USE_SIDPLAY, false) AM_CONDITIONAL(USE_SIDPLAY, false)
AM_CONDITIONAL(USE_TWOLAME, false) AM_CONDITIONAL(USE_TWOLAME, false)
AM_CONDITIONAL(USE_X264, false) AM_CONDITIONAL(USE_X264, false)
@ -441,7 +432,6 @@ ext/cdio/Makefile
ext/dvdread/Makefile ext/dvdread/Makefile
ext/lame/Makefile ext/lame/Makefile
ext/mpeg2dec/Makefile ext/mpeg2dec/Makefile
ext/mpg123/Makefile
ext/sidplay/Makefile ext/sidplay/Makefile
ext/twolame/Makefile ext/twolame/Makefile
ext/x264/Makefile ext/x264/Makefile

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@ -23,7 +23,6 @@
<xi:include href="xml/element-amrwbdec.xml" /> <xi:include href="xml/element-amrwbdec.xml" />
<xi:include href="xml/element-cdiocddasrc.xml" /> <xi:include href="xml/element-cdiocddasrc.xml" />
<xi:include href="xml/element-lamemp3enc.xml" /> <xi:include href="xml/element-lamemp3enc.xml" />
<xi:include href="xml/element-mpg123audiodec.xml" />
<xi:include href="xml/element-rademux.xml" /> <xi:include href="xml/element-rademux.xml" />
<xi:include href="xml/element-rmdemux.xml" /> <xi:include href="xml/element-rmdemux.xml" />
<xi:include href="xml/element-rdtmanager.xml" /> <xi:include href="xml/element-rdtmanager.xml" />
@ -47,7 +46,6 @@
<xi:include href="xml/plugin-dvdsub.xml" /> <xi:include href="xml/plugin-dvdsub.xml" />
<xi:include href="xml/plugin-lame.xml" /> <xi:include href="xml/plugin-lame.xml" />
<xi:include href="xml/plugin-mpeg2dec.xml" /> <xi:include href="xml/plugin-mpeg2dec.xml" />
<xi:include href="xml/plugin-mpg123.xml" />
<xi:include href="xml/plugin-realmedia.xml" /> <xi:include href="xml/plugin-realmedia.xml" />
<xi:include href="xml/plugin-siddec.xml" /> <xi:include href="xml/plugin-siddec.xml" />
<xi:include href="xml/plugin-twolame.xml" /> <xi:include href="xml/plugin-twolame.xml" />

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@ -92,20 +92,6 @@ gst_lamemp3enc_get_type
gst_lamemp3enc_register gst_lamemp3enc_register
</SECTION> </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> <SECTION>
<FILE>element-rademux</FILE> <FILE>element-rademux</FILE>
<TITLE>rademux</TITLE> <TITLE>rademux</TITLE>

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@ -14,7 +14,6 @@ GObject
GstAmrnbDec GstAmrnbDec
GstAmrwbDec GstAmrwbDec
GstDvdLpcmDec GstDvdLpcmDec
GstMpg123AudioDec
GstAudioEncoder GstAudioEncoder
GstAmrnbEnc GstAmrnbEnc
GstLameMP3Enc GstLameMP3Enc

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@ -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>1.12.0</version>
<license>LGPL</license>
<source>gst-plugins-ugly</source>
<package>GStreamer Ugly 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>

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@ -40,12 +40,6 @@ else
MPEG2DEC_DIR = MPEG2DEC_DIR =
endif endif
if USE_MPG123
MPG123_DIR=mpg123
else
MPG123_DIR=
endif
if USE_SIDPLAY if USE_SIDPLAY
SIDPLAY_DIR = sidplay SIDPLAY_DIR = sidplay
else else
@ -72,7 +66,6 @@ SUBDIRS = \
$(DVDREAD_DIR) \ $(DVDREAD_DIR) \
$(LAME_DIR) \ $(LAME_DIR) \
$(MPEG2DEC_DIR) \ $(MPEG2DEC_DIR) \
$(MPG123_DIR) \
$(SIDPLAY_DIR) \ $(SIDPLAY_DIR) \
$(TWOLAME_DIR) \ $(TWOLAME_DIR) \
$(X264_DIR) $(X264_DIR)
@ -85,7 +78,6 @@ DIST_SUBDIRS = \
dvdread \ dvdread \
lame \ lame \
mpeg2dec \ mpeg2dec \
mpg123 \
sidplay \ sidplay \
twolame \ twolame \
x264 x264

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@ -5,7 +5,6 @@ subdir('cdio')
subdir('dvdread') subdir('dvdread')
subdir('lame') subdir('lame')
subdir('mpeg2dec') subdir('mpeg2dec')
subdir('mpg123')
subdir('sidplay') subdir('sidplay')
subdir('twolame') subdir('twolame')
subdir('x264') subdir('x264')

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@ -1,11 +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)
noinst_HEADERS = gstmpg123audiodec.h

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@ -1,634 +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);
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)

View file

@ -1,74 +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__
/* 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

View file

@ -1,16 +0,0 @@
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

View file

@ -38,12 +38,6 @@ else
MPEG2DEC = MPEG2DEC =
endif endif
if USE_MPG123
check_mpg123 = elements/mpg123audiodec
else
check_mpg123 =
endif
if USE_X264 if USE_X264
check_x264enc=elements/x264enc check_x264enc=elements/x264enc
else else
@ -62,7 +56,6 @@ check_PROGRAMS = \
$(AMRNB) \ $(AMRNB) \
$(LAME) \ $(LAME) \
$(MPEG2DEC) \ $(MPEG2DEC) \
$(check_mpg123) \
$(check_x264enc) \ $(check_x264enc) \
$(check_xingmux) $(check_xingmux)
@ -86,14 +79,6 @@ SUPPRESSIONS = $(top_srcdir)/common/gst.supp $(srcdir)/gst-plugins-ugly.supp
elements_amrnbenc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS) elements_amrnbenc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS)
elements_amrnbenc_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD) elements_amrnbenc_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
elements_cmmldec_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS)
elements_cmmlenc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS)
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_mpeg2dec_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS) elements_mpeg2dec_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_mpeg2dec_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \ elements_mpeg2dec_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
-lgstvideo-@GST_API_VERSION@ -lgstvideo-@GST_API_VERSION@

View file

@ -1,6 +1,5 @@
amrnbenc amrnbenc
mpeg2dec mpeg2dec
mpg123audiodec
x264enc x264enc
xingmux xingmux
.dirstamp .dirstamp

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)

View file

@ -2,7 +2,6 @@
ugly_tests = [ ugly_tests = [
[ 'elements/amrnbenc', not amrnb_dep.found() ], [ 'elements/amrnbenc', not amrnb_dep.found() ],
[ 'elements/mpeg2dec', not mpeg2_dep.found(), [ gstvideo_dep ] ], [ 'elements/mpeg2dec', not mpeg2_dep.found(), [ gstvideo_dep ] ],
[ 'elements/mpg123audiodec', not mpg123_dep.found() ],
[ 'elements/x264enc', not x264_dep.found() ], [ 'elements/x264enc', not x264_dep.found() ],
[ 'elements/xingmux' ], [ 'elements/xingmux' ],
[ 'generic/states' ], [ 'generic/states' ],