$(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} \
$(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 \
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, [
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)
ext/Makefile
ext/modplug/Makefile
ext/mpeg2enc/Makefile
-ext/mpg123/Makefile
ext/mimic/Makefile
ext/mplex/Makefile
ext/musepack/Makefile
<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" />
</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>
GstMXFDemux
+++ /dev/null
-<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 <dv@pseudoterminal.org></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>
\ No newline at end of file
MPEG2ENC_DIR=
endif
-if USE_MPG123
-MPG123_DIR=mpg123
-else
-MPG123_DIR=
-endif
-
if USE_MIMIC
MIMIC_DIR=mimic
else
$(LIBVISUAL_DIR) \
$(MODPLUG_DIR) \
$(MPEG2ENC_DIR) \
- $(MPG123_DIR) \
$(MIMIC_DIR) \
$(MPLEX_DIR) \
$(MUSEPACK_DIR) \
modplug \
mimic \
mpeg2enc \
- mpg123 \
mplex \
musepack \
nas \
+++ /dev/null
-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
+++ /dev/null
-/* MP3 decoding plugin for GStreamer using the mpg123 library
- * Copyright (C) 2012 Carlos Rafael Giani
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * 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)
+++ /dev/null
-/* MP3 decoding plugin for GStreamer using the mpg123 library
- * Copyright (C) 2012 Carlos Rafael Giani
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#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
%{_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
check_mpeg2enc =
endif
-if USE_MPG123
-check_mpg123 = elements/mpg123audiodec
-else
-check_mpg123 =
-endif
-
if USE_MPLEX
check_mplex = elements/mplex
else
elements/mpegtsmux \
elements/mpegvideoparse \
elements/mpeg4videoparse \
- $(check_mpg123) \
elements/mxfdemux \
elements/mxfmux \
elements/netsim \
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)
mpegvideoparse
mpeg4videoparse
mpegtsmux
-mpg123audiodec
mplex
mssdemux
mxfdemux
mxfmux
neonhttpsrc
+netsim
ofa
opus
pcapparse
+++ /dev/null
-/* 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)
projects / platform / upstream / gstreamer.git / commitdiff