From 9bfac61f9713cef8c528d79aae884d69a41fbebd Mon Sep 17 00:00:00 2001 From: =?utf8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 8 Apr 2011 19:32:31 +0100 Subject: [PATCH] Remove audioparsers plugin, it has been moved to -good --- Makefile.am | 2 + android/aacparse.mk | 48 - android/amrparse.mk | 48 - configure.ac | 2 - docs/plugins/Makefile.am | 6 - docs/plugins/gst-plugins-bad-plugins-docs.sgml | 7 - docs/plugins/gst-plugins-bad-plugins-sections.txt | 84 -- docs/plugins/inspect/plugin-audioparsersbad.xml | 139 --- gst/audioparsers/Makefile.am | 20 - gst/audioparsers/gstaacparse.c | 715 ----------- gst/audioparsers/gstaacparse.h | 109 -- gst/audioparsers/gstac3parse.c | 507 -------- gst/audioparsers/gstac3parse.h | 73 -- gst/audioparsers/gstamrparse.c | 378 ------ gst/audioparsers/gstamrparse.h | 82 -- gst/audioparsers/gstdcaparse.c | 451 ------- gst/audioparsers/gstdcaparse.h | 78 -- gst/audioparsers/gstflacparse.c | 1354 --------------------- gst/audioparsers/gstflacparse.h | 92 -- gst/audioparsers/gstmpegaudioparse.c | 1252 ------------------- gst/audioparsers/gstmpegaudioparse.h | 111 -- gst/audioparsers/plugin.c | 57 - tests/check/Makefile.am | 22 - tests/check/elements/.gitignore | 6 +- tests/check/elements/aacparse.c | 240 ---- tests/check/elements/ac3parse.c | 163 --- tests/check/elements/amrparse.c | 327 ----- tests/check/elements/flacparse.c | 299 ----- tests/check/elements/mpegaudioparse.c | 172 --- 29 files changed, 3 insertions(+), 6841 deletions(-) delete mode 100644 android/aacparse.mk delete mode 100644 android/amrparse.mk delete mode 100644 docs/plugins/inspect/plugin-audioparsersbad.xml delete mode 100644 gst/audioparsers/Makefile.am delete mode 100644 gst/audioparsers/gstaacparse.c delete mode 100644 gst/audioparsers/gstaacparse.h delete mode 100644 gst/audioparsers/gstac3parse.c delete mode 100644 gst/audioparsers/gstac3parse.h delete mode 100644 gst/audioparsers/gstamrparse.c delete mode 100644 gst/audioparsers/gstamrparse.h delete mode 100644 gst/audioparsers/gstdcaparse.c delete mode 100644 gst/audioparsers/gstdcaparse.h delete mode 100644 gst/audioparsers/gstflacparse.c delete mode 100644 gst/audioparsers/gstflacparse.h delete mode 100644 gst/audioparsers/gstmpegaudioparse.c delete mode 100644 gst/audioparsers/gstmpegaudioparse.h delete mode 100644 gst/audioparsers/plugin.c delete mode 100644 tests/check/elements/aacparse.c delete mode 100644 tests/check/elements/ac3parse.c delete mode 100644 tests/check/elements/amrparse.c delete mode 100644 tests/check/elements/flacparse.c delete mode 100644 tests/check/elements/mpegaudioparse.c diff --git a/Makefile.am b/Makefile.am index 4bcaa88..27200c5 100644 --- a/Makefile.am +++ b/Makefile.am @@ -49,6 +49,7 @@ CRUFT_FILES = \ $(top_builddir)/ext/jack/.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_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/imagefreeze/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/selector/.libs/*.{so,dll,DLL,dylib} \ @@ -56,6 +57,7 @@ CRUFT_FILES = \ $(top_builddir)/gst/valve/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/videoparsers/.libs/libgsth263parse* \ $(top_builddir)/sys/oss4/.libs/*.{so,dll,DLL,dylib} \ + $(top_builddir)/tests/check/elements/{aac,ac3,amr,flac,mpegaudio,dca}parse \ $(top_builddir)/tests/check/elements/autocolorspace \ $(top_builddir)/tests/check/elements/capssetter \ $(top_builddir)/tests/check/elements/imagefreeze \ diff --git a/android/aacparse.mk b/android/aacparse.mk deleted file mode 100644 index 67d8233..0000000 --- a/android/aacparse.mk +++ /dev/null @@ -1,48 +0,0 @@ -LOCAL_PATH:= $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_ARM_MODE := arm - -aacparse_LOCAL_SRC_FILES:= \ - gst/aacparse/gstaacparse.c \ - gst/aacparse/gstbaseparse.c - -LOCAL_SRC_FILES:= $(addprefix ../,$(aacparse_LOCAL_SRC_FILES)) - -LOCAL_SHARED_LIBRARIES := \ - libgstreamer-0.10 \ - libgstbase-0.10 \ - libglib-2.0 \ - libgthread-2.0 \ - libgmodule-2.0 \ - libgobject-2.0 \ - libgstinterfaces-0.10 - -LOCAL_MODULE:= libgstaacparse - -LOCAL_C_INCLUDES := \ - $(LOCAL_PATH)/.. \ - $(LOCAL_PATH)/../gst-libs \ - $(LOCAL_PATH) \ - $(TARGET_OUT_HEADERS)/gstreamer-0.10 \ - $(TARGET_OUT_HEADERS)/glib-2.0 \ - $(TARGET_OUT_HEADERS)/glib-2.0/glib \ - external/libxml2/include - -ifeq ($(STECONF_ANDROID_VERSION),"FROYO") -LOCAL_SHARED_LIBRARIES += libicuuc -LOCAL_C_INCLUDES += external/icu4c/common -endif - - -LOCAL_CFLAGS := -DHAVE_CONFIG_H -# -# define LOCAL_PRELINK_MODULE to false to not use pre-link map -# -LOCAL_PRELINK_MODULE := false - -#It's a gstreamer plugins, and it must be installed on ..../lib/gstreamer-0.10 -LOCAL_MODULE_PATH := $(TARGET_OUT)/lib/gstreamer-0.10 - -include $(BUILD_SHARED_LIBRARY) diff --git a/android/amrparse.mk b/android/amrparse.mk deleted file mode 100644 index 183c52d..0000000 --- a/android/amrparse.mk +++ /dev/null @@ -1,48 +0,0 @@ -LOCAL_PATH:= $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_ARM_MODE := arm - -amrparse_LOCAL_SRC_FILES:= \ - gst/amrparse/gstamrparse.c \ - gst/amrparse/gstbaseparse.c - -LOCAL_SRC_FILES:= $(addprefix ../,$(amrparse_LOCAL_SRC_FILES)) - -LOCAL_SHARED_LIBRARIES := \ - libgstreamer-0.10 \ - libgstbase-0.10 \ - libglib-2.0 \ - libgthread-2.0 \ - libgmodule-2.0 \ - libgobject-2.0 - -LOCAL_MODULE:= libgstamrparse - -LOCAL_C_INCLUDES := \ - $(LOCAL_PATH)/../ext/amrwbenc \ - $(LOCAL_PATH)/.. \ - $(LOCAL_PATH)/../gst-libs \ - $(LOCAL_PATH) \ - $(TARGET_OUT_HEADERS)/gstreamer-0.10 \ - $(TARGET_OUT_HEADERS)/glib-2.0 \ - $(TARGET_OUT_HEADERS)/glib-2.0/glib \ - external/libxml2/include - -ifeq ($(STECONF_ANDROID_VERSION),"FROYO") -LOCAL_SHARED_LIBRARIES += libicuuc -LOCAL_C_INCLUDES += external/icu4c/common -endif - -LOCAL_CFLAGS := -DHAVE_CONFIG_H -# -# define LOCAL_PRELINK_MODULE to false to not use pre-link map -# -LOCAL_PRELINK_MODULE := false - -#It's a gstreamer plugins, and it must be installed on ..../lib/gstreamer-0.10 -LOCAL_MODULE_PATH := $(TARGET_OUT)/lib/gstreamer-0.10 - -include $(BUILD_SHARED_LIBRARY) - diff --git a/configure.ac b/configure.ac index 5fb1c3a..d454f08 100644 --- a/configure.ac +++ b/configure.ac @@ -291,7 +291,6 @@ AG_GST_CHECK_PLUGIN(adpcmdec) AG_GST_CHECK_PLUGIN(adpcmenc) AG_GST_CHECK_PLUGIN(aiff) AG_GST_CHECK_PLUGIN(asfmux) -AG_GST_CHECK_PLUGIN(audioparsers) AG_GST_CHECK_PLUGIN(autoconvert) AG_GST_CHECK_PLUGIN(bayer) AG_GST_CHECK_PLUGIN(camerabin) @@ -1742,7 +1741,6 @@ gst/adpcmdec/Makefile gst/adpcmenc/Makefile gst/aiff/Makefile gst/asfmux/Makefile -gst/audioparsers/Makefile gst/autoconvert/Makefile gst/bayer/Makefile gst/camerabin/Makefile diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 1efa6ad..a939357 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -136,12 +136,6 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/zbar/gstzbar.h \ $(top_srcdir)/gst/aiff/aiffparse.h \ $(top_srcdir)/gst/aiff/aiffmux.h \ - $(top_srcdir)/gst/audioparsers/gstaacparse.h \ - $(top_srcdir)/gst/audioparsers/gstac3parse.h \ - $(top_srcdir)/gst/audioparsers/gstamrparse.h \ - $(top_srcdir)/gst/audioparsers/gstflacparse.h \ - $(top_srcdir)/gst/audioparsers/gstdcaparse.h \ - $(top_srcdir)/gst/audioparsers/gstmpegaudioparse.h \ $(top_srcdir)/gst/autoconvert/gstautoconvert.h \ $(top_srcdir)/gst/camerabin/gstcamerabin.h \ $(top_srcdir)/gst/coloreffects/gstcoloreffects.h \ diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml index c3e1455..fd393a7 100644 --- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml @@ -17,11 +17,8 @@ gst-plugins-bad Elements - - - @@ -42,7 +39,6 @@ - @@ -65,7 +61,6 @@ - @@ -84,7 +79,6 @@ - @@ -135,7 +129,6 @@ gst-plugins-bad Plugins - diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 4a2af52..aa7a81d 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -1,32 +1,4 @@
-element-aacparse -aacparse -GstAacParse - -GstAacParseClass -GST_AACPARSE -GST_AACPARSE_CLASS -GST_IS_AACPARSE -GST_IS_AACPARSE_CLASS -GST_TYPE_AACPARSE -gst_aacparse_get_type -
- -
-element-ac3parse -ac3parse -GstAc3Parse - -GstAc3ParseClass -GST_AC3_PARSE -GST_AC3_PARSE_CLASS -GST_IS_AC3_PARSE -GST_IS_AC3_PARSE_CLASS -GST_TYPE_AC3_PARSE -gst_ac3_parse_get_type -
- -
element-aiffmux aiffmux GstAiffMux @@ -56,20 +28,6 @@ gst_aiff_parse_get_type
-element-amrparse -amrparse -GstAmrParse - -GstAmrParseClass -GST_AMRPARSE -GST_AMRPARSE_CLASS -GST_IS_AMRPARSE -GST_IS_AMRPARSE_CLASS -GST_TYPE_AMRPARSE -gst_amrparse_get_type -
- -
element-amrwbenc amrwbenc GstAmrwbEnc @@ -378,20 +336,6 @@ gst_dc1394_get_type
-element-dcaparse -dcaparse -GstDCAParse - -GstDCAParseClass -GST_DCA_PARSE -GST_DCA_PARSE_CLASS -GST_IS_DCA_PARSE -GST_IS_DCA_PARSE_CLASS -GST_TYPE_DCA_PARSE -gst_dca_parse_get_type -
- -
element-dccpclientsink dccpclientsink GstDCCPClientSink @@ -752,20 +696,6 @@ gst_fisheye_plugin_init
-element-flacparse -flacparse -GstFlacParse - -GstFlacParseClass -GST_FLAC_PARSE -GST_FLAC_PARSE_CLASS -GST_IS_FLAC_PARSE -GST_IS_FLAC_PARSE_CLASS -GST_TYPE_FLAC_PARSE -gst_flac_parse_get_type -
- -
element-fpsdisplaysink fpsdisplaysink GstFPSDisplaySink @@ -1035,20 +965,6 @@ gst_modplug_get_type
-element-mpegaudioparse -mpegaudioparse -GstMpegAudioParse - -GstMpegAudioParseClass -GST_MPEG_AUDIO_PARSE -GST_MPEG_AUDIO_PARSE_CLASS -GST_IS_MPEG_AUDIO_PARSE -GST_IS_MPEG_AUDIO_PARSE_CLASS -GST_TYPE_MPEG_AUDIO_PARSE -gst_mpeg_audio_parse_get_type -
- -
element-mpeg2enc mpeg2enc GstMpeg2enc diff --git a/docs/plugins/inspect/plugin-audioparsersbad.xml b/docs/plugins/inspect/plugin-audioparsersbad.xml deleted file mode 100644 index eb908af..0000000 --- a/docs/plugins/inspect/plugin-audioparsersbad.xml +++ /dev/null @@ -1,139 +0,0 @@ - - audioparsersbad - audioparsers - ../../gst/audioparsers/.libs/libgstaudioparsersbad.so - libgstaudioparsersbad.so - 0.10.21.1 - LGPL - gst-plugins-bad - GStreamer Bad Plug-ins git - Unknown package origin - - - aacparse - AAC audio stream parser - Codec/Parser/Audio - Advanced Audio Coding parser - Stefan Kost <stefan.kost@nokia.com> - - - sink - sink - always -
audio/mpeg, framed=(boolean)false, mpegversion=(int){ 2, 4 }
- - - src - source - always -
audio/mpeg, framed=(boolean)true, mpegversion=(int){ 2, 4 }, stream-format=(string){ raw, adts, adif }
- - - - - ac3parse - AC3 audio stream parser - Codec/Parser/Audio - AC3 parser - Tim-Philipp Müller <tim centricular net> - - - sink - sink - always -
audio/x-ac3, framed=(boolean)false; audio/x-eac3, framed=(boolean)false; audio/ac3, framed=(boolean)false
- - - src - source - always -
audio/x-ac3, framed=(boolean)true, channels=(int)[ 1, 6 ], rate=(int)[ 32000, 48000 ]; audio/x-eac3, framed=(boolean)true, channels=(int)[ 1, 6 ], rate=(int)[ 32000, 48000 ]
- - - - - amrparse - AMR audio stream parser - Codec/Parser/Audio - Adaptive Multi-Rate audio parser - Ronald Bultje <rbultje@ronald.bitfreak.net> - - - sink - sink - always -
audio/x-amr-nb-sh; audio/x-amr-wb-sh
- - - src - source - always -
audio/AMR, rate=(int)8000, channels=(int)1; audio/AMR-WB, rate=(int)16000, channels=(int)1
- - - - - dcaparse - DTS Coherent Acoustics audio stream parser - Codec/Parser/Audio - DCA parser - Tim-Philipp Müller <tim centricular net> - - - sink - sink - always -
audio/x-dts, framed=(boolean)false
- - - src - source - always -
audio/x-dts, framed=(boolean)true, channels=(int)[ 1, 8 ], rate=(int)[ 8000, 192000 ]
- - - - - flacparse - FLAC audio parser - Codec/Parser/Audio - Parses audio with the FLAC lossless audio codec - Sebastian Dröge <sebastian.droege@collabora.co.uk> - - - sink - sink - always -
audio/x-flac, framed=(boolean)false
- - - src - source - always -
audio/x-flac, framed=(boolean)true, channels=(int)[ 1, 8 ], rate=(int)[ 1, 655350 ]
- - - - - mpegaudioparse - MPEG1 Audio Parser - Codec/Parser/Audio - Parses and frames mpeg1 audio streams (levels 1-3), provides seek - Jan Schmidt <thaytan@mad.scientist.com>,Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> - - - sink - sink - always -
audio/mpeg, mpegversion=(int)1, parsed=(boolean)false
- - - src - source - always -
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ], parsed=(boolean)true
- - - - - \ No newline at end of file diff --git a/gst/audioparsers/Makefile.am b/gst/audioparsers/Makefile.am deleted file mode 100644 index 77039c7..0000000 --- a/gst/audioparsers/Makefile.am +++ /dev/null @@ -1,20 +0,0 @@ -plugin_LTLIBRARIES = libgstaudioparsersbad.la - -libgstaudioparsersbad_la_SOURCES = \ - gstaacparse.c gstamrparse.c gstac3parse.c \ - gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \ - plugin.c - -libgstaudioparsersbad_la_CFLAGS = \ - -I$(top_srcdir)/gst-libs \ - $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) -libgstaudioparsersbad_la_LIBADD = \ - $(top_builddir)/gst-libs/gst/baseparse/libgstbaseparse-$(GST_MAJORMINOR).la \ - $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \ - -lgstaudio-$(GST_MAJORMINOR) \ - $(GST_BASE_LIBS) $(GST_LIBS) -libgstaudioparsersbad_la_LDFLAGS = $(PACKAGE_LIBS) $(GST_PLUGIN_LDFLAGS) -libgstaudioparsersbad_la_LIBTOOLFLAGS = --tag=disable-static - -noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \ - gstdcaparse.h gstflacparse.h gstmpegaudioparse.h diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c deleted file mode 100644 index 09e3e71..0000000 --- a/gst/audioparsers/gstaacparse.c +++ /dev/null @@ -1,715 +0,0 @@ -/* GStreamer AAC parser plugin - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-aacparse - * @short_description: AAC parser - * @see_also: #GstAmrParse - * - * This is an AAC parser which handles both ADIF and ADTS stream formats. - * - * As ADIF format is not framed, it is not seekable and stream duration cannot - * be determined either. However, ADTS format AAC clips can be seeked, and parser - * can also estimate playback position and clip duration. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink - * ]| - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstaacparse.h" - - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " - "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, " - "stream-format = (string) { raw, adts, adif };")); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " - "framed = (boolean) false, " "mpegversion = (int) { 2, 4 };")); - -GST_DEBUG_CATEGORY_STATIC (gst_aacparse_debug); -#define GST_CAT_DEFAULT gst_aacparse_debug - - -#define ADIF_MAX_SIZE 40 /* Should be enough */ -#define ADTS_MAX_SIZE 10 /* Should be enough */ - - -#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec) - -gboolean gst_aacparse_start (GstBaseParse * parse); -gboolean gst_aacparse_stop (GstBaseParse * parse); - -static gboolean gst_aacparse_sink_setcaps (GstBaseParse * parse, - GstCaps * caps); - -gboolean gst_aacparse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * size, gint * skipsize); - -GstFlowReturn gst_aacparse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); - -gboolean gst_aacparse_convert (GstBaseParse * parse, - GstFormat src_format, - gint64 src_value, GstFormat dest_format, gint64 * dest_value); - -gint gst_aacparse_get_frame_overhead (GstBaseParse * parse, GstBuffer * buffer); - -gboolean gst_aacparse_event (GstBaseParse * parse, GstEvent * event); - -#define _do_init(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_aacparse_debug, "aacparse", 0, \ - "AAC audio stream parser"); - -GST_BOILERPLATE_FULL (GstAacParse, gst_aacparse, GstBaseParse, - GST_TYPE_BASE_PARSE, _do_init); - -static inline gint -gst_aacparse_get_sample_rate_from_index (guint sr_idx) -{ - static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100, - 32000, 24000, 22050, 16000, 12000, 11025, 8000 - }; - - if (sr_idx < G_N_ELEMENTS (aac_sample_rates)) - return aac_sample_rates[sr_idx]; - GST_WARNING ("Invalid sample rate index %u", sr_idx); - return 0; -} - -/** - * gst_aacparse_base_init: - * @klass: #GstElementClass. - * - */ -static void -gst_aacparse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - - gst_element_class_set_details_simple (element_class, - "AAC audio stream parser", "Codec/Parser/Audio", - "Advanced Audio Coding parser", "Stefan Kost "); -} - - -/** - * gst_aacparse_class_init: - * @klass: #GstAacParseClass. - * - */ -static void -gst_aacparse_class_init (GstAacParseClass * klass) -{ - GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); - - parse_class->start = GST_DEBUG_FUNCPTR (gst_aacparse_start); - parse_class->stop = GST_DEBUG_FUNCPTR (gst_aacparse_stop); - parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aacparse_sink_setcaps); - parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aacparse_parse_frame); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_aacparse_check_valid_frame); -} - - -/** - * gst_aacparse_init: - * @aacparse: #GstAacParse. - * @klass: #GstAacParseClass. - * - */ -static void -gst_aacparse_init (GstAacParse * aacparse, GstAacParseClass * klass) -{ - GST_DEBUG ("initialized"); -} - - -/** - * gst_aacparse_set_src_caps: - * @aacparse: #GstAacParse. - * @sink_caps: (proposed) caps of sink pad - * - * Set source pad caps according to current knowledge about the - * audio stream. - * - * Returns: TRUE if caps were successfully set. - */ -static gboolean -gst_aacparse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps) -{ - GstStructure *s; - GstCaps *src_caps = NULL; - gboolean res = FALSE; - const gchar *stream_format; - - GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps); - if (sink_caps) - src_caps = gst_caps_copy (sink_caps); - else - src_caps = gst_caps_new_simple ("audio/mpeg", NULL); - - gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE, - "mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL); - - switch (aacparse->header_type) { - case DSPAAC_HEADER_NONE: - stream_format = "raw"; - break; - case DSPAAC_HEADER_ADTS: - stream_format = "adts"; - break; - case DSPAAC_HEADER_ADIF: - stream_format = "adif"; - break; - default: - stream_format = NULL; - } - - s = gst_caps_get_structure (src_caps, 0); - if (aacparse->sample_rate > 0) - gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL); - if (aacparse->channels > 0) - gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL); - if (stream_format) - gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL); - - GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps); - - res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps); - gst_caps_unref (src_caps); - return res; -} - - -/** - * gst_aacparse_sink_setcaps: - * @sinkpad: GstPad - * @caps: GstCaps - * - * Implementation of "set_sink_caps" vmethod in #GstBaseParse class. - * - * Returns: TRUE on success. - */ -static gboolean -gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps) -{ - GstAacParse *aacparse; - GstStructure *structure; - gchar *caps_str; - const GValue *value; - - aacparse = GST_AACPARSE (parse); - structure = gst_caps_get_structure (caps, 0); - caps_str = gst_caps_to_string (caps); - - GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str); - g_free (caps_str); - - /* This is needed at least in case of RTP - * Parses the codec_data information to get ObjectType, - * number of channels and samplerate */ - value = gst_structure_get_value (structure, "codec_data"); - if (value) { - GstBuffer *buf = gst_value_get_buffer (value); - - if (buf) { - const guint8 *buffer = GST_BUFFER_DATA (buf); - guint sr_idx; - - sr_idx = ((buffer[0] & 0x07) << 1) | ((buffer[1] & 0x80) >> 7); - aacparse->object_type = (buffer[0] & 0xf8) >> 3; - aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx); - aacparse->channels = (buffer[1] & 0x78) >> 3; - aacparse->header_type = DSPAAC_HEADER_NONE; - aacparse->mpegversion = 4; - - GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d", - aacparse->object_type, aacparse->sample_rate, aacparse->channels); - - /* arrange for metadata and get out of the way */ - gst_aacparse_set_src_caps (aacparse, caps); - gst_base_parse_set_format (parse, - GST_BASE_PARSE_FORMAT_PASSTHROUGH, TRUE); - } else - return FALSE; - - /* caps info overrides */ - gst_structure_get_int (structure, "rate", &aacparse->sample_rate); - gst_structure_get_int (structure, "channels", &aacparse->channels); - } - - return TRUE; -} - - -/** - * gst_aacparse_adts_get_frame_len: - * @data: block of data containing an ADTS header. - * - * This function calculates ADTS frame length from the given header. - * - * Returns: size of the ADTS frame. - */ -static inline guint -gst_aacparse_adts_get_frame_len (const guint8 * data) -{ - return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5); -} - - -/** - * gst_aacparse_check_adts_frame: - * @aacparse: #GstAacParse. - * @data: Data to be checked. - * @avail: Amount of data passed. - * @framesize: If valid ADTS frame was found, this will be set to tell the - * found frame size in bytes. - * @needed_data: If frame was not found, this may be set to tell how much - * more data is needed in the next round to detect the frame - * reliably. This may happen when a frame header candidate - * is found but it cannot be guaranteed to be the header without - * peeking the following data. - * - * Check if the given data contains contains ADTS frame. The algorithm - * will examine ADTS frame header and calculate the frame size. Also, another - * consecutive ADTS frame header need to be present after the found frame. - * Otherwise the data is not considered as a valid ADTS frame. However, this - * "extra check" is omitted when EOS has been received. In this case it is - * enough when data[0] contains a valid ADTS header. - * - * This function may set the #needed_data to indicate that a possible frame - * candidate has been found, but more data (#needed_data bytes) is needed to - * be absolutely sure. When this situation occurs, FALSE will be returned. - * - * When a valid frame is detected, this function will use - * gst_base_parse_set_min_frame_size() function from #GstBaseParse class - * to set the needed bytes for next frame.This way next data chunk is already - * of correct size. - * - * Returns: TRUE if the given data contains a valid ADTS header. - */ -static gboolean -gst_aacparse_check_adts_frame (GstAacParse * aacparse, - const guint8 * data, const guint avail, gboolean drain, - guint * framesize, guint * needed_data) -{ - if (G_UNLIKELY (avail < 2)) - return FALSE; - - if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) { - *framesize = gst_aacparse_adts_get_frame_len (data); - - /* In EOS mode this is enough. No need to examine the data further */ - if (drain) { - return TRUE; - } - - if (*framesize + ADTS_MAX_SIZE > avail) { - /* We have found a possible frame header candidate, but can't be - sure since we don't have enough data to check the next frame */ - GST_DEBUG ("NEED MORE DATA: we need %d, available %d", - *framesize + ADTS_MAX_SIZE, avail); - *needed_data = *framesize + ADTS_MAX_SIZE; - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), - *framesize + ADTS_MAX_SIZE); - return FALSE; - } - - if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) { - guint nextlen = gst_aacparse_adts_get_frame_len (data + (*framesize)); - - GST_LOG ("ADTS frame found, len: %d bytes", *framesize); - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), - nextlen + ADTS_MAX_SIZE); - return TRUE; - } - } - return FALSE; -} - -/* caller ensure sufficient data */ -static inline void -gst_aacparse_parse_adts_header (GstAacParse * aacparse, const guint8 * data, - gint * rate, gint * channels, gint * object, gint * version) -{ - - if (rate) { - gint sr_idx = (data[2] & 0x3c) >> 2; - - *rate = gst_aacparse_get_sample_rate_from_index (sr_idx); - } - if (channels) - *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6); - - if (version) - *version = (data[1] & 0x08) ? 2 : 4; - if (object) - *object = (data[2] & 0xc0) >> 6; -} - -/** - * gst_aacparse_detect_stream: - * @aacparse: #GstAacParse. - * @data: A block of data that needs to be examined for stream characteristics. - * @avail: Size of the given datablock. - * @framesize: If valid stream was found, this will be set to tell the - * first frame size in bytes. - * @skipsize: If valid stream was found, this will be set to tell the first - * audio frame position within the given data. - * - * Examines the given piece of data and try to detect the format of it. It - * checks for "ADIF" header (in the beginning of the clip) and ADTS frame - * header. If the stream is detected, TRUE will be returned and #framesize - * is set to indicate the found frame size. Additionally, #skipsize might - * be set to indicate the number of bytes that need to be skipped, a.k.a. the - * position of the frame inside given data chunk. - * - * Returns: TRUE on success. - */ -static gboolean -gst_aacparse_detect_stream (GstAacParse * aacparse, - const guint8 * data, const guint avail, gboolean drain, - guint * framesize, gint * skipsize) -{ - gboolean found = FALSE; - guint need_data = 0; - guint i = 0; - - GST_DEBUG_OBJECT (aacparse, "Parsing header data"); - - /* FIXME: No need to check for ADIF if we are not in the beginning of the - stream */ - - /* Can we even parse the header? */ - if (avail < ADTS_MAX_SIZE) - return FALSE; - - for (i = 0; i < avail - 4; i++) { - if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) || - strncmp ((char *) data + i, "ADIF", 4) == 0) { - found = TRUE; - - if (i) { - /* Trick: tell the parent class that we didn't find the frame yet, - but make it skip 'i' amount of bytes. Next time we arrive - here we have full frame in the beginning of the data. */ - *skipsize = i; - return FALSE; - } - break; - } - } - if (!found) { - if (i) - *skipsize = i; - return FALSE; - } - - if (gst_aacparse_check_adts_frame (aacparse, data, avail, drain, - framesize, &need_data)) { - gint rate, channels; - - GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize); - - aacparse->header_type = DSPAAC_HEADER_ADTS; - gst_aacparse_parse_adts_header (aacparse, data, &rate, &channels, - &aacparse->object_type, &aacparse->mpegversion); - - gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse), - rate, 1024, 2, 2); - - GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d", - rate, channels, aacparse->object_type, aacparse->mpegversion); - - return TRUE; - } else if (need_data) { - /* This tells the parent class not to skip any data */ - *skipsize = 0; - return FALSE; - } - - if (avail < ADIF_MAX_SIZE) - return FALSE; - - if (memcmp (data + i, "ADIF", 4) == 0) { - const guint8 *adif; - int skip_size = 0; - int bitstream_type; - int sr_idx; - - aacparse->header_type = DSPAAC_HEADER_ADIF; - aacparse->mpegversion = 4; - - /* no way to seek this */ - gst_base_parse_set_seek (GST_BASE_PARSE (aacparse), - GST_BASE_PARSE_SEEK_NONE, 0); - - /* Skip the "ADIF" bytes */ - adif = data + i + 4; - - /* copyright string */ - if (adif[0] & 0x80) - skip_size += 9; /* skip 9 bytes */ - - bitstream_type = adif[0 + skip_size] & 0x10; - aacparse->bitrate = - ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) | - ((unsigned int) adif[1 + skip_size] << 11) | - ((unsigned int) adif[2 + skip_size] << 3) | - ((unsigned int) adif[3 + skip_size] & 0xe0); - - /* CBR */ - if (bitstream_type == 0) { -#if 0 - /* Buffer fullness parsing. Currently not needed... */ - guint num_elems = 0; - guint fullness = 0; - - num_elems = (adif[3 + skip_size] & 0x1e); - GST_INFO ("ADIF num_config_elems: %d", num_elems); - - fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) | - ((unsigned int) adif[4 + skip_size] << 11) | - ((unsigned int) adif[5 + skip_size] << 3) | - ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5); - - GST_INFO ("ADIF buffer fullness: %d", fullness); -#endif - aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) | - ((adif[7 + skip_size] & 0x80) >> 7); - sr_idx = (adif[7 + skip_size] & 0x78) >> 3; - } - /* VBR */ - else { - aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3; - sr_idx = ((adif[4 + skip_size] & 0x07) << 1) | - ((adif[5 + skip_size] & 0x80) >> 7); - } - - /* FIXME: This gives totally wrong results. Duration calculation cannot - be based on this */ - aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx); - - /* baseparse is not given any fps, - * so it will give up on timestamps, seeking, etc */ - - /* FIXME: Can we assume this? */ - aacparse->channels = 2; - - GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d", - aacparse->bitrate, aacparse->sample_rate, aacparse->object_type); - - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512); - - /* arrange for metadata and get out of the way */ - gst_aacparse_set_src_caps (aacparse, - GST_PAD_CAPS (GST_BASE_PARSE_SINK_PAD (aacparse))); - gst_base_parse_set_format (GST_BASE_PARSE (aacparse), - GST_BASE_PARSE_FORMAT_PASSTHROUGH, TRUE); - - *framesize = avail; - return TRUE; - } - - /* This should never happen */ - return FALSE; -} - - -/** - * gst_aacparse_check_valid_frame: - * @parse: #GstBaseParse. - * @buffer: #GstBuffer. - * @framesize: If the buffer contains a valid frame, its size will be put here - * @skipsize: How much data parent class should skip in order to find the - * frame header. - * - * Implementation of "check_valid_frame" vmethod in #GstBaseParse class. - * - * Returns: TRUE if buffer contains a valid frame. - */ -gboolean -gst_aacparse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - const guint8 *data; - GstAacParse *aacparse; - gboolean ret = FALSE; - gboolean sync; - GstBuffer *buffer; - - aacparse = GST_AACPARSE (parse); - buffer = frame->buffer; - data = GST_BUFFER_DATA (buffer); - - sync = GST_BASE_PARSE_FRAME_SYNC (frame); - - if (aacparse->header_type == DSPAAC_HEADER_ADIF || - aacparse->header_type == DSPAAC_HEADER_NONE) { - /* There is nothing to parse */ - *framesize = GST_BUFFER_SIZE (buffer); - ret = TRUE; - - } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || sync == FALSE) { - - ret = gst_aacparse_detect_stream (aacparse, data, GST_BUFFER_SIZE (buffer), - GST_BASE_PARSE_FRAME_DRAIN (frame), framesize, skipsize); - - } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) { - guint needed_data = 1024; - - ret = gst_aacparse_check_adts_frame (aacparse, data, - GST_BUFFER_SIZE (buffer), GST_BASE_PARSE_FRAME_DRAIN (frame), - framesize, &needed_data); - - if (!ret) { - GST_DEBUG ("buffer didn't contain valid frame"); - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), - needed_data); - } - - } else { - GST_DEBUG ("buffer didn't contain valid frame"); - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024); - } - - return ret; -} - - -/** - * gst_aacparse_parse_frame: - * @parse: #GstBaseParse. - * @buffer: #GstBuffer. - * - * Implementation of "parse_frame" vmethod in #GstBaseParse class. - * - * Also determines frame overhead. - * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have - * a per-frame header. - * - * We're making a couple of simplifying assumptions: - * - * 1. We count Program Configuration Elements rather than searching for them - * in the streams to discount them - the overhead is negligible. - * - * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16 - * bits, which should still not be significant enough to warrant the - * additional parsing through the headers - * - * Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed - * forward. Otherwise appropriate error is returned. - */ -GstFlowReturn -gst_aacparse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstAacParse *aacparse; - GstBuffer *buffer; - GstFlowReturn ret = GST_FLOW_OK; - gint rate, channels; - - aacparse = GST_AACPARSE (parse); - buffer = frame->buffer; - - if (G_UNLIKELY (aacparse->header_type != DSPAAC_HEADER_ADTS)) - return ret; - - /* see above */ - frame->overhead = 7; - - gst_aacparse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer), - &rate, &channels, NULL, NULL); - GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels); - - if (G_UNLIKELY (rate != aacparse->sample_rate - || channels != aacparse->channels)) { - aacparse->sample_rate = rate; - aacparse->channels = channels; - - if (!gst_aacparse_set_src_caps (aacparse, - GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) { - /* If linking fails, we need to return appropriate error */ - ret = GST_FLOW_NOT_LINKED; - } - - gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse), - aacparse->sample_rate, 1024, 2, 2); - } - - return ret; -} - - -/** - * gst_aacparse_start: - * @parse: #GstBaseParse. - * - * Implementation of "start" vmethod in #GstBaseParse class. - * - * Returns: TRUE if startup succeeded. - */ -gboolean -gst_aacparse_start (GstBaseParse * parse) -{ - GstAacParse *aacparse; - - aacparse = GST_AACPARSE (parse); - GST_DEBUG ("start"); - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024); - return TRUE; -} - - -/** - * gst_aacparse_stop: - * @parse: #GstBaseParse. - * - * Implementation of "stop" vmethod in #GstBaseParse class. - * - * Returns: TRUE is stopping succeeded. - */ -gboolean -gst_aacparse_stop (GstBaseParse * parse) -{ - GST_DEBUG ("stop"); - return TRUE; -} diff --git a/gst/audioparsers/gstaacparse.h b/gst/audioparsers/gstaacparse.h deleted file mode 100644 index e62bf65..0000000 --- a/gst/audioparsers/gstaacparse.h +++ /dev/null @@ -1,109 +0,0 @@ -/* GStreamer AAC parser - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_AACPARSE_H__ -#define __GST_AACPARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AACPARSE \ - (gst_aacparse_get_type()) -#define GST_AACPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AACPARSE, GstAacParse)) -#define GST_AACPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AACPARSE, GstAacParseClass)) -#define GST_IS_AACPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AACPARSE)) -#define GST_IS_AACPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AACPARSE)) - - -/** - * GstAacHeaderType: - * @DSPAAC_HEADER_NOT_PARSED: Header not parsed yet. - * @DSPAAC_HEADER_UNKNOWN: Unknown (not recognized) header. - * @DSPAAC_HEADER_ADIF: ADIF header found. - * @DSPAAC_HEADER_ADTS: ADTS header found. - * @DSPAAC_HEADER_NONE: Raw stream, no header. - * - * Type header enumeration set in #header_type. - */ -typedef enum { - DSPAAC_HEADER_NOT_PARSED, - DSPAAC_HEADER_UNKNOWN, - DSPAAC_HEADER_ADIF, - DSPAAC_HEADER_ADTS, - DSPAAC_HEADER_NONE -} GstAacHeaderType; - - -typedef struct _GstAacParse GstAacParse; -typedef struct _GstAacParseClass GstAacParseClass; - -/** - * GstAacParse: - * @element: the parent element. - * @object_type: AAC object type of the stream. - * @bitrate: Current media bitrate. - * @sample_rate: Current media samplerate. - * @channels: Current media channel count. - * @frames_per_sec: FPS value of the current stream. - * @header_type: #GstAacHeaderType indicating the current stream type. - * @framecount: The amount of frames that has been processed this far. - * @bytecount: The amount of bytes that has been processed this far. - * @sync: Tells whether the parser is in sync (a.k.a. not searching for header) - * @eos: End-of-Stream indicator. Set when EOS event arrives. - * @duration: Duration of the current stream. - * @ts: Current stream timestamp. - * - * The opaque GstAacParse data structure. - */ -struct _GstAacParse { - GstBaseParse element; - - /* Stream type -related info */ - gint object_type; - gint bitrate; - gint sample_rate; - gint channels; - gint mpegversion; - - GstAacHeaderType header_type; -}; - -/** - * GstAacParseClass: - * @parent_class: Element parent class. - * - * The opaque GstAacParseClass data structure. - */ -struct _GstAacParseClass { - GstBaseParseClass parent_class; -}; - -GType gst_aacparse_get_type (void); - -G_END_DECLS - -#endif /* __GST_AACPARSE_H__ */ diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c deleted file mode 100644 index e001bc3..0000000 --- a/gst/audioparsers/gstac3parse.c +++ /dev/null @@ -1,507 +0,0 @@ -/* GStreamer AC3 parser - * Copyright (C) 2009 Tim-Philipp Müller - * Copyright (C) 2009 Mark Nauwelaerts - * Copyright (C) 2009 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ -/** - * SECTION:element-ac3parse - * @short_description: AC3 parser - * @see_also: #GstAmrParse, #GstAACParse - * - * This is an AC3 parser. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioresample ! audioconvert ! autoaudiosink - * ]| - * - */ - -/* TODO: - * - add support for audio/x-private1-ac3 as well - * - should accept framed and unframed input (needs decodebin fixes first) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstac3parse.h" -#include -#include - -GST_DEBUG_CATEGORY_STATIC (ac3_parse_debug); -#define GST_CAT_DEFAULT ac3_parse_debug - -static const struct -{ - const guint bit_rate; /* nominal bit rate */ - const guint frame_size[3]; /* frame size for 32kHz, 44kHz, and 48kHz */ -} frmsizcod_table[38] = { - { - 32, { - 64, 69, 96}}, { - 32, { - 64, 70, 96}}, { - 40, { - 80, 87, 120}}, { - 40, { - 80, 88, 120}}, { - 48, { - 96, 104, 144}}, { - 48, { - 96, 105, 144}}, { - 56, { - 112, 121, 168}}, { - 56, { - 112, 122, 168}}, { - 64, { - 128, 139, 192}}, { - 64, { - 128, 140, 192}}, { - 80, { - 160, 174, 240}}, { - 80, { - 160, 175, 240}}, { - 96, { - 192, 208, 288}}, { - 96, { - 192, 209, 288}}, { - 112, { - 224, 243, 336}}, { - 112, { - 224, 244, 336}}, { - 128, { - 256, 278, 384}}, { - 128, { - 256, 279, 384}}, { - 160, { - 320, 348, 480}}, { - 160, { - 320, 349, 480}}, { - 192, { - 384, 417, 576}}, { - 192, { - 384, 418, 576}}, { - 224, { - 448, 487, 672}}, { - 224, { - 448, 488, 672}}, { - 256, { - 512, 557, 768}}, { - 256, { - 512, 558, 768}}, { - 320, { - 640, 696, 960}}, { - 320, { - 640, 697, 960}}, { - 384, { - 768, 835, 1152}}, { - 384, { - 768, 836, 1152}}, { - 448, { - 896, 975, 1344}}, { - 448, { - 896, 976, 1344}}, { - 512, { - 1024, 1114, 1536}}, { - 512, { - 1024, 1115, 1536}}, { - 576, { - 1152, 1253, 1728}}, { - 576, { - 1152, 1254, 1728}}, { - 640, { - 1280, 1393, 1920}}, { - 640, { - 1280, 1394, 1920}} -}; - -static const guint fscod_rates[4] = { 48000, 44100, 32000, 0 }; -static const guint acmod_chans[8] = { 2, 1, 2, 3, 3, 4, 4, 5 }; -static const guint numblks[4] = { 1, 2, 3, 6 }; - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-ac3, framed = (boolean) true, " - " channels = (int) [ 1, 6 ], rate = (int) [ 32000, 48000 ]; " - "audio/x-eac3, framed = (boolean) true, " - " channels = (int) [ 1, 6 ], rate = (int) [ 32000, 48000 ] ")); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-ac3, framed = (boolean) false; " - "audio/x-eac3, framed = (boolean) false; " - "audio/ac3, framed = (boolean) false ")); - -static void gst_ac3_parse_finalize (GObject * object); - -static gboolean gst_ac3_parse_start (GstBaseParse * parse); -static gboolean gst_ac3_parse_stop (GstBaseParse * parse); -static gboolean gst_ac3_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * size, gint * skipsize); -static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); - -GST_BOILERPLATE (GstAc3Parse, gst_ac3_parse, GstBaseParse, GST_TYPE_BASE_PARSE); - -static void -gst_ac3_parse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - - gst_element_class_set_details_simple (element_class, - "AC3 audio stream parser", "Codec/Parser/Audio", - "AC3 parser", "Tim-Philipp Müller "); -} - -static void -gst_ac3_parse_class_init (GstAc3ParseClass * klass) -{ - GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); - GObjectClass *object_class = G_OBJECT_CLASS (klass); - - GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0, - "AC3 audio stream parser"); - - object_class->finalize = gst_ac3_parse_finalize; - - parse_class->start = GST_DEBUG_FUNCPTR (gst_ac3_parse_start); - parse_class->stop = GST_DEBUG_FUNCPTR (gst_ac3_parse_stop); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_ac3_parse_check_valid_frame); - parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_parse_frame); -} - -static void -gst_ac3_parse_reset (GstAc3Parse * ac3parse) -{ - ac3parse->channels = -1; - ac3parse->sample_rate = -1; - ac3parse->eac = FALSE; -} - -static void -gst_ac3_parse_init (GstAc3Parse * ac3parse, GstAc3ParseClass * klass) -{ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (ac3parse), 64 * 2); - gst_ac3_parse_reset (ac3parse); -} - -static void -gst_ac3_parse_finalize (GObject * object) -{ - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_ac3_parse_start (GstBaseParse * parse) -{ - GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); - - GST_DEBUG_OBJECT (parse, "starting"); - - gst_ac3_parse_reset (ac3parse); - - return TRUE; -} - -static gboolean -gst_ac3_parse_stop (GstBaseParse * parse) -{ - GST_DEBUG_OBJECT (parse, "stopping"); - - return TRUE; -} - -static gboolean -gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf, - guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid) -{ - GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf); - guint8 fscod, frmsizcod, bsid, bsmod, acmod, lfe_on; - - GST_LOG_OBJECT (ac3parse, "parsing ac3"); - - gst_bit_reader_skip_unchecked (&bits, 16 + 16); - fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); - frmsizcod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 6); - - if (G_UNLIKELY (fscod == 3 || frmsizcod >= G_N_ELEMENTS (frmsizcod_table))) { - GST_DEBUG_OBJECT (ac3parse, "bad fscod=%d frmsizcod=%d", fscod, frmsizcod); - return FALSE; - } - - bsid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 5); - bsmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); - acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); - - /* spec not quite clear here: decoder should decode if less than 8, - * but seemingly only defines 6 and 8 cases */ - if (bsid > 8) { - GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid); - return FALSE; - } else if (bsid != 8 && bsid != 6) { - GST_DEBUG_OBJECT (ac3parse, "undefined bsid=%d", bsid); - } - - if ((acmod & 0x1) && (acmod != 0x1)) /* 3 front channels */ - gst_bit_reader_skip_unchecked (&bits, 2); - if ((acmod & 0x4)) /* if a surround channel exists */ - gst_bit_reader_skip_unchecked (&bits, 2); - if (acmod == 0x2) /* if in 2/0 mode */ - gst_bit_reader_skip_unchecked (&bits, 2); - - lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1); - - if (frame_size) - *frame_size = frmsizcod_table[frmsizcod].frame_size[fscod] * 2; - if (rate) - *rate = fscod_rates[fscod]; - if (chans) - *chans = acmod_chans[acmod] + lfe_on; - if (blks) - *blks = 6; - if (sid) - *sid = 0; - - return TRUE; -} - -static gboolean -gst_ac3_parse_frame_header_eac3 (GstAc3Parse * ac3parse, GstBuffer * buf, - guint * frame_size, guint * rate, guint * chans, guint * blks, guint * sid) -{ - GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf); - guint16 frmsiz, sample_rate, blocks; - guint8 strmtyp, fscod, fscod2, acmod, lfe_on, strmid, numblkscod; - - GST_LOG_OBJECT (ac3parse, "parsing e-ac3"); - - gst_bit_reader_skip_unchecked (&bits, 16); - strmtyp = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* strmtyp */ - if (G_UNLIKELY (strmtyp == 3)) { - GST_DEBUG_OBJECT (ac3parse, "bad strmtyp %d", strmtyp); - return FALSE; - } - - strmid = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* substreamid */ - frmsiz = gst_bit_reader_get_bits_uint16_unchecked (&bits, 11); /* frmsiz */ - fscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod */ - if (fscod == 3) { - fscod2 = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* fscod2 */ - if (G_UNLIKELY (fscod2 == 3)) { - GST_DEBUG_OBJECT (ac3parse, "invalid fscod2"); - return FALSE; - } - sample_rate = fscod_rates[fscod2] / 2; - blocks = 6; - } else { - numblkscod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 2); /* numblkscod */ - sample_rate = fscod_rates[fscod]; - blocks = numblks[numblkscod]; - } - - acmod = gst_bit_reader_get_bits_uint8_unchecked (&bits, 3); /* acmod */ - lfe_on = gst_bit_reader_get_bits_uint8_unchecked (&bits, 1); /* lfeon */ - - gst_bit_reader_skip_unchecked (&bits, 5); /* bsid */ - - if (frame_size) - *frame_size = (frmsiz + 1) * 2; - if (rate) - *rate = sample_rate; - if (chans) - *chans = acmod_chans[acmod] + lfe_on; - if (blks) - *blks = blocks; - if (sid) - *sid = (strmtyp & 0x1) << 3 | strmid; - - return TRUE; -} - -static gboolean -gst_ac3_parse_frame_header (GstAc3Parse * parse, GstBuffer * buf, - guint * framesize, guint * rate, guint * chans, guint * blocks, - guint * sid, gboolean * eac) -{ - GstBitReader bits = GST_BIT_READER_INIT_FROM_BUFFER (buf); - guint16 sync; - guint8 bsid; - - GST_MEMDUMP_OBJECT (parse, "AC3 frame sync", GST_BUFFER_DATA (buf), 16); - - sync = gst_bit_reader_get_bits_uint16_unchecked (&bits, 16); - gst_bit_reader_skip_unchecked (&bits, 16 + 8); - bsid = gst_bit_reader_peek_bits_uint8_unchecked (&bits, 5); - - if (G_UNLIKELY (sync != 0x0b77)) - return FALSE; - - GST_LOG_OBJECT (parse, "bsid = %d", bsid); - - if (bsid <= 10) { - if (eac) - *eac = FALSE; - return gst_ac3_parse_frame_header_ac3 (parse, buf, framesize, rate, chans, - blocks, sid); - } else if (bsid <= 16) { - if (eac) - *eac = TRUE; - return gst_ac3_parse_frame_header_eac3 (parse, buf, framesize, rate, chans, - blocks, sid); - } else { - GST_DEBUG_OBJECT (parse, "unexpected bsid %d", bsid); - return FALSE; - } -} - -static gboolean -gst_ac3_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); - GstBuffer *buf = frame->buffer; - GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - gint off; - gboolean sync, drain; - - if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6)) - return FALSE; - - off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000, - 0, GST_BUFFER_SIZE (buf)); - - GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); - - /* didn't find anything that looks like a sync word, skip */ - if (off < 0) { - *skipsize = GST_BUFFER_SIZE (buf) - 3; - return FALSE; - } - - /* possible frame header, but not at offset 0? skip bytes before sync */ - if (off > 0) { - *skipsize = off; - return FALSE; - } - - /* make sure the values in the frame header look sane */ - if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, NULL, NULL, - NULL, NULL, NULL)) { - *skipsize = off + 2; - return FALSE; - } - - GST_LOG_OBJECT (parse, "got frame"); - - sync = GST_BASE_PARSE_FRAME_SYNC (frame); - drain = GST_BASE_PARSE_FRAME_DRAIN (frame); - - if (!sync && !drain) { - guint16 word = 0; - - GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword"); - - if (!gst_byte_reader_skip (&reader, *framesize) || - !gst_byte_reader_get_uint16_be (&reader, &word)) { - GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data"); - gst_base_parse_set_min_frame_size (parse, *framesize + 6); - *skipsize = 0; - return FALSE; - } else { - if (word != 0x0b77) { - GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word); - *skipsize = off + 2; - return FALSE; - } else { - /* ok, got sync now, let's assume constant frame size */ - gst_base_parse_set_min_frame_size (parse, *framesize); - } - } - } - - return TRUE; -} - -static GstFlowReturn -gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); - GstBuffer *buf = frame->buffer; - guint fsize, rate, chans, blocks, sid; - gboolean eac; - - if (!gst_ac3_parse_frame_header (ac3parse, buf, &fsize, &rate, &chans, - &blocks, &sid, &eac)) - goto broken_header; - - GST_LOG_OBJECT (parse, "size: %u, rate: %u, chans: %u", fsize, rate, chans); - - if (G_UNLIKELY (sid)) { - /* dependent frame, no need to (ac)count for or consider further */ - GST_LOG_OBJECT (parse, "sid: %d", sid); - frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME; - /* TODO maybe also mark as DELTA_UNIT, - * if that does not surprise baseparse elsewhere */ - /* occupies same time space as previous base frame */ - if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf))) - GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf); - /* only return if we already arranged for caps */ - if (G_LIKELY (ac3parse->sample_rate > 0)) - return GST_FLOW_OK; - } - - if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans - || ac3parse->eac != ac3parse->eac)) { - GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3", - "framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate, - "channels", G_TYPE_INT, chans, NULL); - gst_buffer_set_caps (buf, caps); - gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); - gst_caps_unref (caps); - - ac3parse->sample_rate = rate; - ac3parse->channels = chans; - ac3parse->eac = eac; - - gst_base_parse_set_frame_props (parse, rate, 256 * blocks, 2, 2); - } - - return GST_FLOW_OK; - -/* ERRORS */ -broken_header: - { - /* this really shouldn't ever happen */ - GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL)); - return GST_FLOW_ERROR; - } -} diff --git a/gst/audioparsers/gstac3parse.h b/gst/audioparsers/gstac3parse.h deleted file mode 100644 index 781554b..0000000 --- a/gst/audioparsers/gstac3parse.h +++ /dev/null @@ -1,73 +0,0 @@ -/* GStreamer AC3 parser - * Copyright (C) 2009 Tim-Philipp Müller - * Copyright (C) 2009 Mark Nauwelaerts - * Copyright (C) 2009 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_AC3_PARSE_H__ -#define __GST_AC3_PARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AC3_PARSE \ - (gst_ac3_parse_get_type()) -#define GST_AC3_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AC3_PARSE, GstAc3Parse)) -#define GST_AC3_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AC3_PARSE, GstAc3ParseClass)) -#define GST_IS_AC3_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AC3_PARSE)) -#define GST_IS_AC3_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AC3_PARSE)) - -typedef struct _GstAc3Parse GstAc3Parse; -typedef struct _GstAc3ParseClass GstAc3ParseClass; - -/** - * GstAc3Parse: - * - * The opaque GstAc3Parse object - */ -struct _GstAc3Parse { - GstBaseParse baseparse; - - /*< private >*/ - gint sample_rate; - gint channels; - gboolean eac; -}; - -/** - * GstAc3ParseClass: - * @parent_class: Element parent class. - * - * The opaque GstAc3ParseClass data structure. - */ -struct _GstAc3ParseClass { - GstBaseParseClass baseparse_class; -}; - -GType gst_ac3_parse_get_type (void); - -G_END_DECLS - -#endif /* __GST_AC3_PARSE_H__ */ diff --git a/gst/audioparsers/gstamrparse.c b/gst/audioparsers/gstamrparse.c deleted file mode 100644 index 42481a2..0000000 --- a/gst/audioparsers/gstamrparse.c +++ /dev/null @@ -1,378 +0,0 @@ -/* GStreamer Adaptive Multi-Rate parser plugin - * Copyright (C) 2006 Edgard Lima - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-amrparse - * @short_description: AMR parser - * @see_also: #GstAmrnbDec, #GstAmrnbEnc - * - * This is an AMR parser capable of handling both narrow-band and wideband - * formats. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink - * ]| - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstamrparse.h" - - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1;" - "audio/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1;") - ); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-amr-nb-sh; audio/x-amr-wb-sh")); - -GST_DEBUG_CATEGORY_STATIC (gst_amrparse_debug); -#define GST_CAT_DEFAULT gst_amrparse_debug - -static const gint block_size_nb[16] = - { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 }; - -static const gint block_size_wb[16] = - { 17, 23, 32, 36, 40, 46, 50, 58, 60, 5, -1, -1, -1, -1, 0, 0 }; - -/* AMR has a "hardcoded" framerate of 50fps */ -#define AMR_FRAMES_PER_SECOND 50 -#define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND) -#define AMR_MIME_HEADER_SIZE 9 - -gboolean gst_amrparse_start (GstBaseParse * parse); -gboolean gst_amrparse_stop (GstBaseParse * parse); - -static gboolean gst_amrparse_sink_setcaps (GstBaseParse * parse, - GstCaps * caps); - -gboolean gst_amrparse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize); - -GstFlowReturn gst_amrparse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); - -#define _do_init(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_amrparse_debug, "amrparse", 0, \ - "AMR-NB audio stream parser"); - -GST_BOILERPLATE_FULL (GstAmrParse, gst_amrparse, GstBaseParse, - GST_TYPE_BASE_PARSE, _do_init); - - -/** - * gst_amrparse_base_init: - * @klass: #GstElementClass. - * - */ -static void -gst_amrparse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - - gst_element_class_set_details_simple (element_class, - "AMR audio stream parser", "Codec/Parser/Audio", - "Adaptive Multi-Rate audio parser", - "Ronald Bultje "); -} - - -/** - * gst_amrparse_class_init: - * @klass: GstAmrParseClass. - * - */ -static void -gst_amrparse_class_init (GstAmrParseClass * klass) -{ - GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); - - parse_class->start = GST_DEBUG_FUNCPTR (gst_amrparse_start); - parse_class->stop = GST_DEBUG_FUNCPTR (gst_amrparse_stop); - parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amrparse_sink_setcaps); - parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_amrparse_parse_frame); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_amrparse_check_valid_frame); -} - - -/** - * gst_amrparse_init: - * @amrparse: #GstAmrParse - * @klass: #GstAmrParseClass. - * - */ -static void -gst_amrparse_init (GstAmrParse * amrparse, GstAmrParseClass * klass) -{ - /* init rest */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62); - GST_DEBUG ("initialized"); - -} - - -/** - * gst_amrparse_set_src_caps: - * @amrparse: #GstAmrParse. - * - * Set source pad caps according to current knowledge about the - * audio stream. - * - * Returns: TRUE if caps were successfully set. - */ -static gboolean -gst_amrparse_set_src_caps (GstAmrParse * amrparse) -{ - GstCaps *src_caps = NULL; - gboolean res = FALSE; - - if (amrparse->wide) { - GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-WB"); - src_caps = gst_caps_new_simple ("audio/AMR-WB", - "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 16000, NULL); - } else { - GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-NB"); - /* Max. size of NB frame is 31 bytes, so we can set the min. frame - size to 32 (+1 for next frame header) */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 32); - src_caps = gst_caps_new_simple ("audio/AMR", - "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL); - } - gst_pad_use_fixed_caps (GST_BASE_PARSE (amrparse)->srcpad); - res = gst_pad_set_caps (GST_BASE_PARSE (amrparse)->srcpad, src_caps); - gst_caps_unref (src_caps); - return res; -} - - -/** - * gst_amrparse_sink_setcaps: - * @sinkpad: GstPad - * @caps: GstCaps - * - * Returns: TRUE on success. - */ -static gboolean -gst_amrparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps) -{ - GstAmrParse *amrparse; - GstStructure *structure; - const gchar *name; - - amrparse = GST_AMRPARSE (parse); - structure = gst_caps_get_structure (caps, 0); - name = gst_structure_get_name (structure); - - GST_DEBUG_OBJECT (amrparse, "setcaps: %s", name); - - if (!strncmp (name, "audio/x-amr-wb-sh", 17)) { - amrparse->block_size = block_size_wb; - amrparse->wide = 1; - } else if (!strncmp (name, "audio/x-amr-nb-sh", 17)) { - amrparse->block_size = block_size_nb; - amrparse->wide = 0; - } else { - GST_WARNING ("Unknown caps"); - return FALSE; - } - - amrparse->need_header = FALSE; - gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); - gst_amrparse_set_src_caps (amrparse); - return TRUE; -} - -/** - * gst_amrparse_parse_header: - * @amrparse: #GstAmrParse - * @data: Header data to be parsed. - * @skipsize: Output argument where the frame size will be stored. - * - * Check if the given data contains an AMR mime header. - * - * Returns: TRUE on success. - */ -static gboolean -gst_amrparse_parse_header (GstAmrParse * amrparse, - const guint8 * data, gint * skipsize) -{ - GST_DEBUG_OBJECT (amrparse, "Parsing header data"); - - if (!memcmp (data, "#!AMR-WB\n", 9)) { - GST_DEBUG_OBJECT (amrparse, "AMR-WB detected"); - amrparse->block_size = block_size_wb; - amrparse->wide = TRUE; - *skipsize = amrparse->header = 9; - } else if (!memcmp (data, "#!AMR\n", 6)) { - GST_DEBUG_OBJECT (amrparse, "AMR-NB detected"); - amrparse->block_size = block_size_nb; - amrparse->wide = FALSE; - *skipsize = amrparse->header = 6; - } else - return FALSE; - - gst_amrparse_set_src_caps (amrparse); - return TRUE; -} - - -/** - * gst_amrparse_check_valid_frame: - * @parse: #GstBaseParse. - * @buffer: #GstBuffer. - * @framesize: Output variable where the found frame size is put. - * @skipsize: Output variable which tells how much data needs to be skipped - * until a frame header is found. - * - * Implementation of "check_valid_frame" vmethod in #GstBaseParse class. - * - * Returns: TRUE if the given data contains valid frame. - */ -gboolean -gst_amrparse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - GstBuffer *buffer; - const guint8 *data; - gint fsize, mode, dsize; - GstAmrParse *amrparse; - - amrparse = GST_AMRPARSE (parse); - buffer = frame->buffer; - data = GST_BUFFER_DATA (buffer); - dsize = GST_BUFFER_SIZE (buffer); - - GST_LOG ("buffer: %d bytes", dsize); - - if (amrparse->need_header) { - if (dsize >= AMR_MIME_HEADER_SIZE && - gst_amrparse_parse_header (amrparse, data, skipsize)) { - amrparse->need_header = FALSE; - gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); - } else { - GST_WARNING ("media doesn't look like a AMR format"); - } - /* We return FALSE, so this frame won't get pushed forward. Instead, - the "skip" value is set, so next time we will receive a valid frame. */ - return FALSE; - } - - /* Does this look like a possible frame header candidate? */ - if ((data[0] & 0x83) == 0) { - /* Yep. Retrieve the frame size */ - mode = (data[0] >> 3) & 0x0F; - fsize = amrparse->block_size[mode] + 1; /* +1 for the header byte */ - - /* We recognize this data as a valid frame when: - * - We are in sync. There is no need for extra checks then - * - We are in EOS. There might not be enough data to check next frame - * - Sync is lost, but the following data after this frame seem - * to contain a valid header as well (and there is enough data to - * perform this check) - */ - if (fsize && - (GST_BASE_PARSE_FRAME_SYNC (frame) || GST_BASE_PARSE_FRAME_DRAIN (frame) - || (dsize > fsize && (data[fsize] & 0x83) == 0))) { - *framesize = fsize; - return TRUE; - } - } - - GST_LOG ("sync lost"); - return FALSE; -} - - -/** - * gst_amrparse_parse_frame: - * @parse: #GstBaseParse. - * @buffer: #GstBuffer. - * - * Implementation of "parse" vmethod in #GstBaseParse class. - * - * Returns: #GstFlowReturn defining the parsing status. - */ -GstFlowReturn -gst_amrparse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - return GST_FLOW_OK; -} - - -/** - * gst_amrparse_start: - * @parse: #GstBaseParse. - * - * Implementation of "start" vmethod in #GstBaseParse class. - * - * Returns: TRUE on success. - */ -gboolean -gst_amrparse_start (GstBaseParse * parse) -{ - GstAmrParse *amrparse; - - amrparse = GST_AMRPARSE (parse); - GST_DEBUG ("start"); - amrparse->need_header = TRUE; - amrparse->header = 0; - return TRUE; -} - - -/** - * gst_amrparse_stop: - * @parse: #GstBaseParse. - * - * Implementation of "stop" vmethod in #GstBaseParse class. - * - * Returns: TRUE on success. - */ -gboolean -gst_amrparse_stop (GstBaseParse * parse) -{ - GstAmrParse *amrparse; - - amrparse = GST_AMRPARSE (parse); - GST_DEBUG ("stop"); - amrparse->need_header = TRUE; - amrparse->header = 0; - return TRUE; -} diff --git a/gst/audioparsers/gstamrparse.h b/gst/audioparsers/gstamrparse.h deleted file mode 100644 index 04cd6e7..0000000 --- a/gst/audioparsers/gstamrparse.h +++ /dev/null @@ -1,82 +0,0 @@ -/* GStreamer Adaptive Multi-Rate parser - * Copyright (C) 2004 Ronald Bultje - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_AMRPARSE_H__ -#define __GST_AMRPARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AMRPARSE \ - (gst_amrparse_get_type()) -#define GST_AMRPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRPARSE, GstAmrParse)) -#define GST_AMRPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRPARSE, GstAmrParseClass)) -#define GST_IS_AMRPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRPARSE)) -#define GST_IS_AMRPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRPARSE)) - - -typedef struct _GstAmrParse GstAmrParse; -typedef struct _GstAmrParseClass GstAmrParseClass; - -/** - * GstAmrParse: - * @element: the parent element. - * @block_size: Pointer to frame size lookup table. - * @need_header: Tells whether the MIME header should be read in the beginning. - * @wide: Wideband mode. - * @eos: Indicates the EOS situation. Set when EOS event is received. - * @sync: Tells whether the parser is in sync. - * @framecount: Total amount of frames handled. - * @bytecount: Total amount of bytes handled. - * @ts: Timestamp of the current media. - * - * The opaque GstAacParse data structure. - */ -struct _GstAmrParse { - GstBaseParse element; - const gint *block_size; - gboolean need_header; - gint header; - gboolean wide; -}; - -/** - * GstAmrParseClass: - * @parent_class: Element parent class. - * - * The opaque GstAmrParseClass data structure. - */ -struct _GstAmrParseClass { - GstBaseParseClass parent_class; -}; - -GType gst_amrparse_get_type (void); - -G_END_DECLS - -#endif /* __GST_AMRPARSE_H__ */ diff --git a/gst/audioparsers/gstdcaparse.c b/gst/audioparsers/gstdcaparse.c deleted file mode 100644 index 7e478e4..0000000 --- a/gst/audioparsers/gstdcaparse.c +++ /dev/null @@ -1,451 +0,0 @@ -/* GStreamer DCA parser - * Copyright (C) 2010 Tim-Philipp Müller - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-dcaparse - * @short_description: DCA (DTS Coherent Acoustics) parser - * @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse - * - * This is a DCA (DTS Coherent Acoustics) parser. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink - * ]| - * - */ - -/* TODO: - * - should accept framed and unframed input (needs decodebin fixes first) - * - seeking in raw .dts files doesn't seem to work, but duration estimate ok - * - * - if frames have 'odd' durations, the frame durations (plus timestamps) - * aren't adjusted up occasionally to make up for rounding error gaps. - * (e.g. if 512 samples per frame @ 48kHz = 10.666666667 ms/frame) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstdcaparse.h" -#include -#include - -GST_DEBUG_CATEGORY_STATIC (dca_parse_debug); -#define GST_CAT_DEFAULT dca_parse_debug - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-dts," - " framed = (boolean) true," - " channels = (int) [ 1, 8 ]," - " rate = (int) [ 8000, 192000 ]," - " depth = (int) { 14, 16 }," - " endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }")); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-dts, framed = (boolean) false")); - -static void gst_dca_parse_finalize (GObject * object); - -static gboolean gst_dca_parse_start (GstBaseParse * parse); -static gboolean gst_dca_parse_stop (GstBaseParse * parse); -static gboolean gst_dca_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * size, gint * skipsize); -static GstFlowReturn gst_dca_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); - -GST_BOILERPLATE (GstDcaParse, gst_dca_parse, GstBaseParse, GST_TYPE_BASE_PARSE); - -static void -gst_dca_parse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - - gst_element_class_set_details_simple (element_class, - "DTS Coherent Acoustics audio stream parser", "Codec/Parser/Audio", - "DCA parser", "Tim-Philipp Müller "); -} - -static void -gst_dca_parse_class_init (GstDcaParseClass * klass) -{ - GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); - GObjectClass *object_class = G_OBJECT_CLASS (klass); - - GST_DEBUG_CATEGORY_INIT (dca_parse_debug, "dcaparse", 0, - "DCA audio stream parser"); - - object_class->finalize = gst_dca_parse_finalize; - - parse_class->start = GST_DEBUG_FUNCPTR (gst_dca_parse_start); - parse_class->stop = GST_DEBUG_FUNCPTR (gst_dca_parse_stop); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_dca_parse_check_valid_frame); - parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_parse_frame); -} - -static void -gst_dca_parse_reset (GstDcaParse * dcaparse) -{ - dcaparse->channels = -1; - dcaparse->rate = -1; - dcaparse->depth = -1; - dcaparse->endianness = -1; - dcaparse->block_size = -1; - dcaparse->frame_size = -1; - dcaparse->last_sync = 0; -} - -static void -gst_dca_parse_init (GstDcaParse * dcaparse, GstDcaParseClass * klass) -{ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (dcaparse), - DCA_MIN_FRAMESIZE); - gst_dca_parse_reset (dcaparse); -} - -static void -gst_dca_parse_finalize (GObject * object) -{ - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_dca_parse_start (GstBaseParse * parse) -{ - GstDcaParse *dcaparse = GST_DCA_PARSE (parse); - - GST_DEBUG_OBJECT (parse, "starting"); - - gst_dca_parse_reset (dcaparse); - - return TRUE; -} - -static gboolean -gst_dca_parse_stop (GstBaseParse * parse) -{ - GST_DEBUG_OBJECT (parse, "stopping"); - - return TRUE; -} - -static gboolean -gst_dca_parse_parse_header (GstDcaParse * dcaparse, - const GstByteReader * reader, guint * frame_size, - guint * sample_rate, guint * channels, guint * depth, - gint * endianness, guint * num_blocks, guint * samples_per_block, - gboolean * terminator) -{ - static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025, - 22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000 - }; - static const guint8 channels_table[16] = { 1, 2, 2, 2, 2, 3, 3, 4, 4, 5, - 6, 6, 6, 7, 8, 8 - }; - GstByteReader r = *reader; - guint16 hdr[8]; - guint32 marker; - guint chans, lfe, i; - - if (gst_byte_reader_get_remaining (&r) < (4 + sizeof (hdr))) - return FALSE; - - marker = gst_byte_reader_peek_uint32_be_unchecked (&r); - - /* raw big endian or 14-bit big endian */ - if (marker == 0x7FFE8001 || marker == 0x1FFFE800) { - for (i = 0; i < G_N_ELEMENTS (hdr); ++i) - hdr[i] = gst_byte_reader_get_uint16_be_unchecked (&r); - } else - /* raw little endian or 14-bit little endian */ - if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) { - for (i = 0; i < G_N_ELEMENTS (hdr); ++i) - hdr[i] = gst_byte_reader_get_uint16_le_unchecked (&r); - } else { - return FALSE; - } - - GST_LOG_OBJECT (dcaparse, "dts sync marker 0x%08x at offset %u", marker, - gst_byte_reader_get_pos (reader)); - - /* 14-bit mode */ - if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) { - if ((hdr[2] & 0xFFF0) != 0x07F0) - return FALSE; - /* discard top 2 bits (2 void), shift in 2 */ - hdr[0] = (hdr[0] << 2) | ((hdr[1] >> 12) & 0x0003); - /* discard top 4 bits (2 void, 2 shifted into hdr[0]), shift in 4 etc. */ - hdr[1] = (hdr[1] << 4) | ((hdr[2] >> 10) & 0x000F); - hdr[2] = (hdr[2] << 6) | ((hdr[3] >> 8) & 0x003F); - hdr[3] = (hdr[3] << 8) | ((hdr[4] >> 6) & 0x00FF); - hdr[4] = (hdr[4] << 10) | ((hdr[5] >> 4) & 0x03FF); - hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF); - hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF); - g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001); - } - - GST_LOG_OBJECT (dcaparse, "frame header: %04x%04x%04x%04x", - hdr[2], hdr[3], hdr[4], hdr[5]); - - *terminator = (hdr[2] & 0x80) ? FALSE : TRUE; - *samples_per_block = ((hdr[2] >> 10) & 0x1f) + 1; - *num_blocks = ((hdr[2] >> 2) & 0x7F) + 1; - *frame_size = (((hdr[2] & 0x03) << 12) | (hdr[3] >> 4)) + 1; - chans = ((hdr[3] & 0x0F) << 2) | (hdr[4] >> 14); - *sample_rate = sample_rates[(hdr[4] >> 10) & 0x0F]; - lfe = (hdr[5] >> 9) & 0x03; - - GST_TRACE_OBJECT (dcaparse, "frame size %u, num_blocks %u, rate %u, " - "samples per block %u", *frame_size, *num_blocks, *sample_rate, - *samples_per_block); - - if (*num_blocks < 6 || *frame_size < 96 || *sample_rate == 0) - return FALSE; - - if (marker == 0x1FFFE800 || marker == 0xFF1F00E8) - *frame_size = (*frame_size * 16) / 14; /* FIXME: round up? */ - - if (chans < G_N_ELEMENTS (channels_table)) - *channels = channels_table[chans] + ((lfe) ? 1 : 0); - else - *channels = 0; - - if (depth) - *depth = (marker == 0x1FFFE800 || marker == 0xFF1F00E8) ? 14 : 16; - if (endianness) - *endianness = (marker == 0xFE7F0180 || marker == 0xFF1F00E8) ? - G_LITTLE_ENDIAN : G_BIG_ENDIAN; - - GST_TRACE_OBJECT (dcaparse, "frame size %u, channels %u, rate %u, " - "num_blocks %u, samples_per_block %u", *frame_size, *channels, - *sample_rate, *num_blocks, *samples_per_block); - - return TRUE; -} - -static gint -gst_dca_parse_find_sync (GstDcaParse * dcaparse, GstByteReader * reader, - const GstBuffer * buf, guint32 * sync) -{ - guint32 best_sync = 0; - guint best_offset = G_MAXUINT; - gint off; - - /* FIXME: verify syncs via _parse_header() here already */ - - /* Raw little endian */ - off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xfe7f0180, - 0, GST_BUFFER_SIZE (buf)); - if (off >= 0 && off < best_offset) { - best_offset = off; - best_sync = 0xfe7f0180; - } - - /* Raw big endian */ - off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x7ffe8001, - 0, GST_BUFFER_SIZE (buf)); - if (off >= 0 && off < best_offset) { - best_offset = off; - best_sync = 0x7ffe8001; - } - - /* FIXME: check next 2 bytes as well for 14-bit formats (but then don't - * forget to adjust the *skipsize= in _check_valid_frame() */ - - /* 14-bit little endian */ - off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0xff1f00e8, - 0, GST_BUFFER_SIZE (buf)); - if (off >= 0 && off < best_offset) { - best_offset = off; - best_sync = 0xff1f00e8; - } - - /* 14-bit big endian */ - off = gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x1fffe800, - 0, GST_BUFFER_SIZE (buf)); - if (off >= 0 && off < best_offset) { - best_offset = off; - best_sync = 0x1fffe800; - } - - if (best_offset == G_MAXUINT) - return -1; - - *sync = best_sync; - return best_offset; -} - -static gboolean -gst_dca_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - GstDcaParse *dcaparse = GST_DCA_PARSE (parse); - GstBuffer *buf = frame->buffer; - GstByteReader r = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - gboolean parser_draining; - gboolean parser_in_sync; - gboolean terminator; - guint32 sync = 0; - guint size, rate, chans, num_blocks, samples_per_block; - gint off = -1; - - if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 16)) - return FALSE; - - parser_in_sync = GST_BASE_PARSE_FRAME_SYNC (frame); - - if (G_LIKELY (parser_in_sync && dcaparse->last_sync != 0)) { - off = gst_byte_reader_masked_scan_uint32 (&r, 0xffffffff, - dcaparse->last_sync, 0, GST_BUFFER_SIZE (buf)); - } - - if (G_UNLIKELY (off < 0)) { - off = gst_dca_parse_find_sync (dcaparse, &r, buf, &sync); - } - - /* didn't find anything that looks like a sync word, skip */ - if (off < 0) { - *skipsize = GST_BUFFER_SIZE (buf) - 3; - GST_DEBUG_OBJECT (dcaparse, "no sync, skipping %d bytes", *skipsize); - return FALSE; - } - - GST_LOG_OBJECT (parse, "possible sync %08x at buffer offset %d", sync, off); - - /* possible frame header, but not at offset 0? skip bytes before sync */ - if (off > 0) { - *skipsize = off; - return FALSE; - } - - /* make sure the values in the frame header look sane */ - if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, NULL, - NULL, &num_blocks, &samples_per_block, &terminator)) { - *skipsize = 4; - return FALSE; - } - - GST_LOG_OBJECT (parse, "got frame, sync %08x, size %u, rate %d, channels %d", - sync, size, rate, chans); - - *framesize = size; - - dcaparse->last_sync = sync; - - parser_draining = GST_BASE_PARSE_FRAME_DRAIN (frame); - - if (!parser_in_sync && !parser_draining) { - /* check for second frame to be sure */ - GST_DEBUG_OBJECT (dcaparse, "resyncing; checking next frame syncword"); - if (GST_BUFFER_SIZE (buf) >= (size + 16)) { - guint s2, r2, c2, n2, s3; - gboolean t; - - GST_MEMDUMP ("buf", GST_BUFFER_DATA (buf), size + 16); - gst_byte_reader_init_from_buffer (&r, buf); - gst_byte_reader_skip_unchecked (&r, size); - - if (!gst_dca_parse_parse_header (dcaparse, &r, &s2, &r2, &c2, NULL, NULL, - &n2, &s3, &t)) { - GST_DEBUG_OBJECT (dcaparse, "didn't find second syncword"); - *skipsize = 4; - return FALSE; - } - - /* ok, got sync now, let's assume constant frame size */ - gst_base_parse_set_min_frame_size (parse, size); - } else { - /* FIXME: baseparse always seems to hand us buffers of min_frame_size - * bytes, which is unhelpful here */ - GST_LOG_OBJECT (dcaparse, "next sync out of reach (%u < %u)", - GST_BUFFER_SIZE (buf), size + 16); - /* *skipsize = 0; */ - /* return FALSE; */ - } - } - - return TRUE; -} - -static GstFlowReturn -gst_dca_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstDcaParse *dcaparse = GST_DCA_PARSE (parse); - GstBuffer *buf = frame->buffer; - GstByteReader r = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - guint size, rate, chans, depth, block_size, num_blocks, samples_per_block; - gint endianness; - gboolean terminator; - - if (!gst_dca_parse_parse_header (dcaparse, &r, &size, &rate, &chans, &depth, - &endianness, &num_blocks, &samples_per_block, &terminator)) - goto broken_header; - - block_size = num_blocks * samples_per_block; - - if (G_UNLIKELY (dcaparse->rate != rate || dcaparse->channels != chans - || dcaparse->depth != depth || dcaparse->endianness != endianness - || (!terminator && dcaparse->block_size != block_size) - || (size != dcaparse->frame_size))) { - GstCaps *caps; - - caps = gst_caps_new_simple ("audio/x-dts", - "framed", G_TYPE_BOOLEAN, TRUE, - "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans, - "endianness", G_TYPE_INT, endianness, "depth", G_TYPE_INT, depth, - "block-size", G_TYPE_INT, block_size, "frame-size", G_TYPE_INT, size, - NULL); - gst_buffer_set_caps (buf, caps); - gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); - gst_caps_unref (caps); - - dcaparse->rate = rate; - dcaparse->channels = chans; - dcaparse->depth = depth; - dcaparse->endianness = endianness; - dcaparse->block_size = block_size; - dcaparse->frame_size = size; - - gst_base_parse_set_frame_props (parse, rate, block_size, 0, 0); - } - - return GST_FLOW_OK; - -/* ERRORS */ -broken_header: - { - /* this really shouldn't ever happen */ - GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL)); - return GST_FLOW_ERROR; - } -} diff --git a/gst/audioparsers/gstdcaparse.h b/gst/audioparsers/gstdcaparse.h deleted file mode 100644 index eefc252..0000000 --- a/gst/audioparsers/gstdcaparse.h +++ /dev/null @@ -1,78 +0,0 @@ -/* GStreamer DCA parser - * Copyright (C) 2010 Tim-Philipp Müller - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_DCA_PARSE_H__ -#define __GST_DCA_PARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_DCA_PARSE \ - (gst_dca_parse_get_type()) -#define GST_DCA_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_DCA_PARSE, GstDcaParse)) -#define GST_DCA_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_DCA_PARSE, GstDcaParseClass)) -#define GST_IS_DCA_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_DCA_PARSE)) -#define GST_IS_DCA_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_DCA_PARSE)) - -#define DCA_MIN_FRAMESIZE 96 -#define DCA_MAX_FRAMESIZE 18725 /* 16384*16/14 */ - -typedef struct _GstDcaParse GstDcaParse; -typedef struct _GstDcaParseClass GstDcaParseClass; - -/** - * GstDcaParse: - * - * The opaque GstDcaParse object - */ -struct _GstDcaParse { - GstBaseParse baseparse; - - /*< private >*/ - gint rate; - gint channels; - gint depth; - gint endianness; - gint block_size; - gint frame_size; - - guint32 last_sync; -}; - -/** - * GstDcaParseClass: - * @parent_class: Element parent class. - * - * The opaque GstDcaParseClass data structure. - */ -struct _GstDcaParseClass { - GstBaseParseClass baseparse_class; -}; - -GType gst_dca_parse_get_type (void); - -G_END_DECLS - -#endif /* __GST_DCA_PARSE_H__ */ diff --git a/gst/audioparsers/gstflacparse.c b/gst/audioparsers/gstflacparse.c deleted file mode 100644 index 8306e8e..0000000 --- a/gst/audioparsers/gstflacparse.c +++ /dev/null @@ -1,1354 +0,0 @@ -/* GStreamer - * - * Copyright (C) 2008 Sebastian Dröge . - * Copyright (C) 2009 Mark Nauwelaerts - * Copyright (C) 2009 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-flacparse - * @see_also: flacdec, oggdemux, vorbisparse - * - * The flacparse element will parse the header packets of the FLAC - * stream and put them as the streamheader in the caps. This is used in the - * multifdsink case where you want to stream live FLAC streams to multiple - * clients, each client has to receive the streamheaders first before they can - * consume the FLAC packets. - * - * This element also makes sure that the buffers that it pushes out are properly - * timestamped and that their offset and offset_end are set. The buffers that - * flacparse outputs have all of the metadata that oggmux expects to receive, - * which allows you to (for example) remux an ogg/flac or convert a native FLAC - * format file to an ogg bitstream. - * - * - * Example pipelines - * |[ - * gst-launch -v filesrc location=sine.flac ! flacparse ! identity \ - * ! oggmux ! filesink location=sine-remuxed.ogg - * ]| This pipeline converts a native FLAC format file to an ogg bitstream. - * It also illustrates that the streamheader is set in the caps, and that each - * buffer has the timestamp, duration, offset, and offset_end set. - * - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstflacparse.h" - -#include -#include -#include - -#include -#include - -GST_DEBUG_CATEGORY_STATIC (flacparse_debug); -#define GST_CAT_DEFAULT flacparse_debug - -/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */ -static const guint8 crc8_table[256] = { - 0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15, - 0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D, - 0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65, - 0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D, - 0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5, - 0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD, - 0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85, - 0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD, - 0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2, - 0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA, - 0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2, - 0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A, - 0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32, - 0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A, - 0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42, - 0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A, - 0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C, - 0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4, - 0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC, - 0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4, - 0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C, - 0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44, - 0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C, - 0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34, - 0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B, - 0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63, - 0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B, - 0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13, - 0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB, - 0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83, - 0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB, - 0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3 -}; - -static guint8 -gst_flac_calculate_crc8 (const guint8 * data, guint length) -{ - guint8 crc = 0; - - while (length--) { - crc = crc8_table[crc ^ *data]; - ++data; - } - - return crc; -} - -/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */ -static const guint16 crc16_table[256] = { - 0x0000, 0x8005, 0x800f, 0x000a, 0x801b, 0x001e, 0x0014, 0x8011, - 0x8033, 0x0036, 0x003c, 0x8039, 0x0028, 0x802d, 0x8027, 0x0022, - 0x8063, 0x0066, 0x006c, 0x8069, 0x0078, 0x807d, 0x8077, 0x0072, - 0x0050, 0x8055, 0x805f, 0x005a, 0x804b, 0x004e, 0x0044, 0x8041, - 0x80c3, 0x00c6, 0x00cc, 0x80c9, 0x00d8, 0x80dd, 0x80d7, 0x00d2, - 0x00f0, 0x80f5, 0x80ff, 0x00fa, 0x80eb, 0x00ee, 0x00e4, 0x80e1, - 0x00a0, 0x80a5, 0x80af, 0x00aa, 0x80bb, 0x00be, 0x00b4, 0x80b1, - 0x8093, 0x0096, 0x009c, 0x8099, 0x0088, 0x808d, 0x8087, 0x0082, - 0x8183, 0x0186, 0x018c, 0x8189, 0x0198, 0x819d, 0x8197, 0x0192, - 0x01b0, 0x81b5, 0x81bf, 0x01ba, 0x81ab, 0x01ae, 0x01a4, 0x81a1, - 0x01e0, 0x81e5, 0x81ef, 0x01ea, 0x81fb, 0x01fe, 0x01f4, 0x81f1, - 0x81d3, 0x01d6, 0x01dc, 0x81d9, 0x01c8, 0x81cd, 0x81c7, 0x01c2, - 0x0140, 0x8145, 0x814f, 0x014a, 0x815b, 0x015e, 0x0154, 0x8151, - 0x8173, 0x0176, 0x017c, 0x8179, 0x0168, 0x816d, 0x8167, 0x0162, - 0x8123, 0x0126, 0x012c, 0x8129, 0x0138, 0x813d, 0x8137, 0x0132, - 0x0110, 0x8115, 0x811f, 0x011a, 0x810b, 0x010e, 0x0104, 0x8101, - 0x8303, 0x0306, 0x030c, 0x8309, 0x0318, 0x831d, 0x8317, 0x0312, - 0x0330, 0x8335, 0x833f, 0x033a, 0x832b, 0x032e, 0x0324, 0x8321, - 0x0360, 0x8365, 0x836f, 0x036a, 0x837b, 0x037e, 0x0374, 0x8371, - 0x8353, 0x0356, 0x035c, 0x8359, 0x0348, 0x834d, 0x8347, 0x0342, - 0x03c0, 0x83c5, 0x83cf, 0x03ca, 0x83db, 0x03de, 0x03d4, 0x83d1, - 0x83f3, 0x03f6, 0x03fc, 0x83f9, 0x03e8, 0x83ed, 0x83e7, 0x03e2, - 0x83a3, 0x03a6, 0x03ac, 0x83a9, 0x03b8, 0x83bd, 0x83b7, 0x03b2, - 0x0390, 0x8395, 0x839f, 0x039a, 0x838b, 0x038e, 0x0384, 0x8381, - 0x0280, 0x8285, 0x828f, 0x028a, 0x829b, 0x029e, 0x0294, 0x8291, - 0x82b3, 0x02b6, 0x02bc, 0x82b9, 0x02a8, 0x82ad, 0x82a7, 0x02a2, - 0x82e3, 0x02e6, 0x02ec, 0x82e9, 0x02f8, 0x82fd, 0x82f7, 0x02f2, - 0x02d0, 0x82d5, 0x82df, 0x02da, 0x82cb, 0x02ce, 0x02c4, 0x82c1, - 0x8243, 0x0246, 0x024c, 0x8249, 0x0258, 0x825d, 0x8257, 0x0252, - 0x0270, 0x8275, 0x827f, 0x027a, 0x826b, 0x026e, 0x0264, 0x8261, - 0x0220, 0x8225, 0x822f, 0x022a, 0x823b, 0x023e, 0x0234, 0x8231, - 0x8213, 0x0216, 0x021c, 0x8219, 0x0208, 0x820d, 0x8207, 0x0202 -}; - -static guint16 -gst_flac_calculate_crc16 (const guint8 * data, guint length) -{ - guint16 crc = 0; - - while (length--) { - crc = ((crc << 8) ^ crc16_table[(crc >> 8) ^ *data]) & 0xffff; - data++; - } - - return crc; -} - -enum -{ - PROP_0, - PROP_CHECK_FRAME_CHECKSUMS -}; - -#define DEFAULT_CHECK_FRAME_CHECKSUMS FALSE - -static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-flac, framed = (boolean) true, " - "channels = (int) [ 1, 8 ], " "rate = (int) [ 1, 655350 ]") - ); - -static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-flac, framed = (boolean) false") - ); - -static void gst_flac_parse_finalize (GObject * object); -static void gst_flac_parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_flac_parse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static gboolean gst_flac_parse_start (GstBaseParse * parse); -static gboolean gst_flac_parse_stop (GstBaseParse * parse); -static gboolean gst_flac_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize); -static GstFlowReturn gst_flac_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); -static GstFlowReturn gst_flac_parse_pre_push_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); - -GST_BOILERPLATE (GstFlacParse, gst_flac_parse, GstBaseParse, - GST_TYPE_BASE_PARSE); - -static void -gst_flac_parse_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_factory)); - - gst_element_class_set_details_simple (element_class, "FLAC audio parser", - "Codec/Parser/Audio", - "Parses audio with the FLAC lossless audio codec", - "Sebastian Dröge "); - - GST_DEBUG_CATEGORY_INIT (flacparse_debug, "flacparse", 0, - "Flac parser element"); -} - -static void -gst_flac_parse_class_init (GstFlacParseClass * klass) -{ - GObjectClass *gobject_class = G_OBJECT_CLASS (klass); - GstBaseParseClass *baseparse_class = GST_BASE_PARSE_CLASS (klass); - - gobject_class->finalize = gst_flac_parse_finalize; - gobject_class->set_property = gst_flac_parse_set_property; - gobject_class->get_property = gst_flac_parse_get_property; - - g_object_class_install_property (gobject_class, PROP_CHECK_FRAME_CHECKSUMS, - g_param_spec_boolean ("check-frame-checksums", "Check Frame Checksums", - "Check the overall checksums of every frame", - DEFAULT_CHECK_FRAME_CHECKSUMS, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - baseparse_class->start = GST_DEBUG_FUNCPTR (gst_flac_parse_start); - baseparse_class->stop = GST_DEBUG_FUNCPTR (gst_flac_parse_stop); - baseparse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_flac_parse_check_valid_frame); - baseparse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_flac_parse_parse_frame); - baseparse_class->pre_push_frame = - GST_DEBUG_FUNCPTR (gst_flac_parse_pre_push_frame); -} - -static void -gst_flac_parse_init (GstFlacParse * flacparse, GstFlacParseClass * klass) -{ - flacparse->check_frame_checksums = DEFAULT_CHECK_FRAME_CHECKSUMS; -} - -static void -gst_flac_parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (object); - - switch (prop_id) { - case PROP_CHECK_FRAME_CHECKSUMS: - flacparse->check_frame_checksums = g_value_get_boolean (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_flac_parse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (object); - - switch (prop_id) { - case PROP_CHECK_FRAME_CHECKSUMS: - g_value_set_boolean (value, flacparse->check_frame_checksums); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_flac_parse_finalize (GObject * object) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (object); - - if (flacparse->tags) { - gst_tag_list_free (flacparse->tags); - flacparse->tags = NULL; - } - - g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL); - g_list_free (flacparse->headers); - flacparse->headers = NULL; - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_flac_parse_start (GstBaseParse * parse) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (parse); - - flacparse->state = GST_FLAC_PARSE_STATE_INIT; - flacparse->min_blocksize = 0; - flacparse->max_blocksize = 0; - flacparse->min_framesize = 0; - flacparse->max_framesize = 0; - - flacparse->upstream_length = -1; - - flacparse->samplerate = 0; - flacparse->channels = 0; - flacparse->bps = 0; - flacparse->total_samples = 0; - - flacparse->offset = GST_CLOCK_TIME_NONE; - flacparse->blocking_strategy = 0; - flacparse->block_size = 0; - flacparse->sample_number = 0; - - /* "fLaC" marker */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 4); - /* inform baseclass we can come up with ts, based on counters in packets */ - gst_base_parse_set_format (GST_BASE_PARSE (flacparse), - GST_BASE_PARSE_FORMAT_HAS_TIME, TRUE); - - return TRUE; -} - -static gboolean -gst_flac_parse_stop (GstBaseParse * parse) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (parse); - - if (flacparse->tags) { - gst_tag_list_free (flacparse->tags); - flacparse->tags = NULL; - } - - g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL); - g_list_free (flacparse->headers); - flacparse->headers = NULL; - - return TRUE; -} - -static const guint8 sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 }; - -static const guint16 blocksize_table[16] = { - 0, 192, 576 << 0, 576 << 1, 576 << 2, 576 << 3, 0, 0, - 256 << 0, 256 << 1, 256 << 2, 256 << 3, 256 << 4, 256 << 5, 256 << 6, - 256 << 7, -}; - -static const guint32 sample_rate_table[16] = { - 0, - 88200, 176400, 192000, - 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, - 0, 0, 0, 0, -}; - -typedef enum -{ - FRAME_HEADER_VALID, - FRAME_HEADER_INVALID, - FRAME_HEADER_MORE_DATA -} FrameHeaderCheckReturn; - -static FrameHeaderCheckReturn -gst_flac_parse_frame_header_is_valid (GstFlacParse * flacparse, - const guint8 * data, guint size, gboolean set, guint16 * block_size_ret) -{ - GstBitReader reader = GST_BIT_READER_INIT (data, size); - guint8 blocking_strategy; - guint16 block_size; - guint32 samplerate = 0; - guint64 sample_number; - guint8 channels, bps; - guint8 tmp = 0; - guint8 actual_crc, expected_crc = 0; - - /* Skip 14 bit sync code */ - gst_bit_reader_skip_unchecked (&reader, 14); - - /* Must be 0 */ - if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) != 0) - goto error; - - /* 0 == fixed block size, 1 == variable block size */ - blocking_strategy = gst_bit_reader_get_bits_uint8_unchecked (&reader, 1); - - /* block size index, calculation of the real blocksize below */ - block_size = gst_bit_reader_get_bits_uint16_unchecked (&reader, 4); - if (block_size == 0) - goto error; - - /* sample rate index, calculation of the real samplerate below */ - samplerate = gst_bit_reader_get_bits_uint16_unchecked (&reader, 4); - if (samplerate == 0x0f) - goto error; - - /* channel assignment */ - channels = gst_bit_reader_get_bits_uint8_unchecked (&reader, 4); - if (channels < 8) { - channels++; - } else if (channels <= 10) { - channels = 2; - } else if (channels > 10) { - goto error; - } - if (flacparse->channels && flacparse->channels != channels) - goto error; - - /* bits per sample */ - bps = gst_bit_reader_get_bits_uint8_unchecked (&reader, 3); - if (bps == 0x03 || bps == 0x07) { - goto error; - } else if (bps == 0 && flacparse->bps == 0) { - goto need_streaminfo; - } - bps = sample_size_table[bps]; - if (flacparse->bps && bps != flacparse->bps) - goto error; - - /* reserved, must be 0 */ - if (gst_bit_reader_get_bits_uint8_unchecked (&reader, 1) != 0) - goto error; - - /* read "utf8" encoded sample/frame number */ - { - gint len = 0; - - len = gst_bit_reader_get_bits_uint8_unchecked (&reader, 8); - - /* This is slightly faster than a loop */ - if (!(len & 0x80)) { - sample_number = len; - len = 0; - } else if ((len & 0xc0) && !(len & 0x20)) { - sample_number = len & 0x1f; - len = 1; - } else if ((len & 0xe0) && !(len & 0x10)) { - sample_number = len & 0x0f; - len = 2; - } else if ((len & 0xf0) && !(len & 0x08)) { - sample_number = len & 0x07; - len = 3; - } else if ((len & 0xf8) && !(len & 0x04)) { - sample_number = len & 0x03; - len = 4; - } else if ((len & 0xfc) && !(len & 0x02)) { - sample_number = len & 0x01; - len = 5; - } else if ((len & 0xfe) && !(len & 0x01)) { - sample_number = len & 0x0; - len = 6; - } else { - goto error; - } - - if ((blocking_strategy == 0 && len > 5) || - (blocking_strategy == 1 && len > 6)) - goto error; - - while (len > 0) { - if (!gst_bit_reader_get_bits_uint8 (&reader, &tmp, 8)) - goto need_more_data; - - if ((tmp & 0xc0) != 0x80) - goto error; - - sample_number <<= 6; - sample_number |= (tmp & 0x3f); - len--; - } - } - - /* calculate real blocksize from the blocksize index */ - if (block_size == 0) { - goto error; - } else if (block_size == 6) { - if (!gst_bit_reader_get_bits_uint16 (&reader, &block_size, 8)) - goto need_more_data; - block_size++; - } else if (block_size == 7) { - if (!gst_bit_reader_get_bits_uint16 (&reader, &block_size, 16)) - goto need_more_data; - block_size++; - } else { - block_size = blocksize_table[block_size]; - } - - /* calculate the real samplerate from the samplerate index */ - if (samplerate == 0 && flacparse->samplerate == 0) { - goto need_streaminfo; - } else if (samplerate < 12) { - samplerate = sample_rate_table[samplerate]; - } else if (samplerate == 12) { - if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 8)) - goto need_more_data; - samplerate *= 1000; - } else if (samplerate == 13) { - if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 16)) - goto need_more_data; - } else if (samplerate == 14) { - if (!gst_bit_reader_get_bits_uint32 (&reader, &samplerate, 16)) - goto need_more_data; - samplerate *= 10; - } - - if (flacparse->samplerate && flacparse->samplerate != samplerate) - goto error; - - /* check crc-8 for the header */ - if (!gst_bit_reader_get_bits_uint8 (&reader, &expected_crc, 8)) - goto need_more_data; - - actual_crc = - gst_flac_calculate_crc8 (data, - (gst_bit_reader_get_pos (&reader) / 8) - 1); - if (actual_crc != expected_crc) - goto error; - - if (set) { - flacparse->block_size = block_size; - if (!flacparse->samplerate) - flacparse->samplerate = samplerate; - if (!flacparse->bps) - flacparse->bps = bps; - if (!flacparse->blocking_strategy) - flacparse->blocking_strategy = blocking_strategy; - if (!flacparse->channels) - flacparse->channels = channels; - if (!flacparse->sample_number) - flacparse->sample_number = sample_number; - - GST_DEBUG_OBJECT (flacparse, - "Parsed frame at offset %" G_GUINT64_FORMAT ":\n" "Block size: %u\n" - "Sample/Frame number: %" G_GUINT64_FORMAT, flacparse->offset, - flacparse->block_size, flacparse->sample_number); - } - - if (block_size_ret) - *block_size_ret = block_size; - - return FRAME_HEADER_VALID; - -need_streaminfo: - GST_ERROR_OBJECT (flacparse, "Need STREAMINFO"); - return FRAME_HEADER_INVALID; -error: - return FRAME_HEADER_INVALID; - -need_more_data: - return FRAME_HEADER_MORE_DATA; -} - -static gboolean -gst_flac_parse_frame_is_valid (GstFlacParse * flacparse, - GstBaseParseFrame * frame, guint * ret) -{ - GstBuffer *buffer; - const guint8 *data; - guint max, size, remaining; - guint i, search_start, search_end; - FrameHeaderCheckReturn header_ret; - guint16 block_size; - - buffer = frame->buffer; - data = GST_BUFFER_DATA (buffer); - size = GST_BUFFER_SIZE (buffer); - - if (size <= flacparse->min_framesize) - goto need_more; - - header_ret = - gst_flac_parse_frame_header_is_valid (flacparse, data, size, TRUE, - &block_size); - if (header_ret == FRAME_HEADER_INVALID) { - *ret = 0; - return FALSE; - } else if (header_ret == FRAME_HEADER_MORE_DATA) { - goto need_more; - } - - /* mind unknown framesize */ - search_start = MAX (2, flacparse->min_framesize); - if (flacparse->max_framesize) - search_end = MIN (size, flacparse->max_framesize + 9 + 2); - else - search_end = size; - search_end -= 2; - - remaining = size; - - for (i = search_start; i < search_end; i++, remaining--) { - if ((GST_READ_UINT16_BE (data + i) & 0xfffe) == 0xfff8) { - header_ret = - gst_flac_parse_frame_header_is_valid (flacparse, data + i, remaining, - FALSE, NULL); - if (header_ret == FRAME_HEADER_VALID) { - if (flacparse->check_frame_checksums) { - guint16 actual_crc = gst_flac_calculate_crc16 (data, i - 2); - guint16 expected_crc = GST_READ_UINT16_BE (data + i - 2); - - if (actual_crc != expected_crc) - continue; - } - *ret = i; - flacparse->block_size = block_size; - return TRUE; - } else if (header_ret == FRAME_HEADER_MORE_DATA) { - goto need_more; - } - } - } - - /* For the last frame output everything to the end */ - if (G_UNLIKELY (GST_BASE_PARSE_FRAME_DRAIN (frame))) { - if (flacparse->check_frame_checksums) { - guint16 actual_crc = gst_flac_calculate_crc16 (data, size - 2); - guint16 expected_crc = GST_READ_UINT16_BE (data + size - 2); - - if (actual_crc == expected_crc) { - *ret = size; - flacparse->block_size = block_size; - return TRUE; - } - } else { - *ret = size; - flacparse->block_size = block_size; - return TRUE; - } - } - -need_more: - max = flacparse->max_framesize + 16; - if (max == 16) - max = 1 << 24; - *ret = MIN (size + 4096, max); - return FALSE; -} - -static gboolean -gst_flac_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (parse); - GstBuffer *buffer = frame->buffer; - const guint8 *data = GST_BUFFER_DATA (buffer); - - if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 4)) - return FALSE; - - if (flacparse->state == GST_FLAC_PARSE_STATE_INIT) { - if (memcmp (GST_BUFFER_DATA (buffer), "fLaC", 4) == 0) { - GST_DEBUG_OBJECT (flacparse, "fLaC marker found"); - *framesize = 4; - return TRUE; - } else if (data[0] == 0xff && (data[1] >> 2) == 0x3e) { - GST_DEBUG_OBJECT (flacparse, "Found headerless FLAC"); - /* Minimal size of a frame header */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 9); - flacparse->state = GST_FLAC_PARSE_STATE_GENERATE_HEADERS; - *skipsize = 0; - return FALSE; - } else { - GST_DEBUG_OBJECT (flacparse, "fLaC marker not found"); - return FALSE; - } - } else if (flacparse->state == GST_FLAC_PARSE_STATE_HEADERS) { - guint size = 4 + ((data[1] << 16) | (data[2] << 8) | (data[3])); - - GST_DEBUG_OBJECT (flacparse, "Found metadata block of size %u", size); - *framesize = size; - return TRUE; - } else { - if ((GST_READ_UINT16_BE (data) & 0xfffe) == 0xfff8) { - gboolean ret; - guint next; - - flacparse->offset = GST_BUFFER_OFFSET (buffer); - flacparse->blocking_strategy = 0; - flacparse->block_size = 0; - flacparse->sample_number = 0; - - GST_DEBUG_OBJECT (flacparse, "Found sync code"); - ret = gst_flac_parse_frame_is_valid (flacparse, frame, &next); - if (ret) { - *framesize = next; - return TRUE; - } else { - /* If we're at EOS and the frame was not valid, drop it! */ - if (G_UNLIKELY (GST_BASE_PARSE_FRAME_DRAIN (frame))) { - GST_WARNING_OBJECT (flacparse, "EOS"); - return FALSE; - } - - if (next == 0) { - } else if (next > GST_BUFFER_SIZE (buffer)) { - GST_DEBUG_OBJECT (flacparse, "Requesting %u bytes", next); - *skipsize = 0; - gst_base_parse_set_min_frame_size (parse, next); - return FALSE; - } else { - GST_ERROR_OBJECT (flacparse, - "Giving up on invalid frame (%d bytes)", - GST_BUFFER_SIZE (buffer)); - return FALSE; - } - } - } else { - GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buffer); - gint off; - - off = - gst_byte_reader_masked_scan_uint32 (&reader, 0xfffc0000, 0xfff80000, - 0, GST_BUFFER_SIZE (buffer)); - - if (off > 0) { - GST_DEBUG_OBJECT (parse, "Possible sync at buffer offset %d", off); - *skipsize = off; - return FALSE; - } else { - GST_DEBUG_OBJECT (flacparse, "Sync code not found"); - *skipsize = GST_BUFFER_SIZE (buffer) - 3; - return FALSE; - } - } - } - - return FALSE; -} - -static gboolean -gst_flac_parse_handle_streaminfo (GstFlacParse * flacparse, GstBuffer * buffer) -{ - GstBitReader reader = GST_BIT_READER_INIT_FROM_BUFFER (buffer); - - if (GST_BUFFER_SIZE (buffer) != 4 + 34) { - GST_ERROR_OBJECT (flacparse, "Invalid metablock size for STREAMINFO: %u", - GST_BUFFER_SIZE (buffer)); - return FALSE; - } - - /* Skip metadata block header */ - gst_bit_reader_skip (&reader, 32); - - if (!gst_bit_reader_get_bits_uint16 (&reader, &flacparse->min_blocksize, 16)) - goto error; - if (flacparse->min_blocksize < 16) { - GST_ERROR_OBJECT (flacparse, "Invalid minimum block size: %u", - flacparse->min_blocksize); - return FALSE; - } - - if (!gst_bit_reader_get_bits_uint16 (&reader, &flacparse->max_blocksize, 16)) - goto error; - if (flacparse->max_blocksize < 16) { - GST_ERROR_OBJECT (flacparse, "Invalid maximum block size: %u", - flacparse->max_blocksize); - return FALSE; - } - - if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->min_framesize, 24)) - goto error; - if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->max_framesize, 24)) - goto error; - - if (!gst_bit_reader_get_bits_uint32 (&reader, &flacparse->samplerate, 20)) - goto error; - if (flacparse->samplerate == 0) { - GST_ERROR_OBJECT (flacparse, "Invalid sample rate 0"); - return FALSE; - } - - if (!gst_bit_reader_get_bits_uint8 (&reader, &flacparse->channels, 3)) - goto error; - flacparse->channels++; - if (flacparse->channels > 8) { - GST_ERROR_OBJECT (flacparse, "Invalid number of channels %u", - flacparse->channels); - return FALSE; - } - - if (!gst_bit_reader_get_bits_uint8 (&reader, &flacparse->bps, 5)) - goto error; - flacparse->bps++; - - if (!gst_bit_reader_get_bits_uint64 (&reader, &flacparse->total_samples, 36)) - goto error; - if (flacparse->total_samples) - gst_base_parse_set_duration (GST_BASE_PARSE (flacparse), GST_FORMAT_TIME, - GST_FRAMES_TO_CLOCK_TIME (flacparse->total_samples, - flacparse->samplerate), 0); - - GST_DEBUG_OBJECT (flacparse, "STREAMINFO:\n" - "\tmin/max blocksize: %u/%u,\n" - "\tmin/max framesize: %u/%u,\n" - "\tsamplerate: %u,\n" - "\tchannels: %u,\n" - "\tbits per sample: %u,\n" - "\ttotal samples: %" G_GUINT64_FORMAT, - flacparse->min_blocksize, flacparse->max_blocksize, - flacparse->min_framesize, flacparse->max_framesize, - flacparse->samplerate, - flacparse->channels, flacparse->bps, flacparse->total_samples); - - return TRUE; - -error: - GST_ERROR_OBJECT (flacparse, "Failed to read data"); - return FALSE; -} - -static gboolean -gst_flac_parse_handle_vorbiscomment (GstFlacParse * flacparse, - GstBuffer * buffer) -{ - flacparse->tags = gst_tag_list_from_vorbiscomment_buffer (buffer, - GST_BUFFER_DATA (buffer), 4, NULL); - - if (flacparse->tags == NULL) { - GST_ERROR_OBJECT (flacparse, "Invalid vorbiscomment block"); - } else if (gst_tag_list_is_empty (flacparse->tags)) { - gst_tag_list_free (flacparse->tags); - flacparse->tags = NULL; - } - - return TRUE; -} - -static gboolean -gst_flac_parse_handle_picture (GstFlacParse * flacparse, GstBuffer * buffer) -{ - GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buffer); - const guint8 *data = GST_BUFFER_DATA (buffer); - guint32 img_len = 0, img_type = 0; - guint32 img_mimetype_len = 0, img_description_len = 0; - - if (!gst_byte_reader_skip (&reader, 4)) - goto error; - - if (!gst_byte_reader_get_uint32_be (&reader, &img_type)) - goto error; - - if (!gst_byte_reader_get_uint32_be (&reader, &img_mimetype_len)) - goto error; - if (!gst_byte_reader_skip (&reader, img_mimetype_len)) - goto error; - - if (!gst_byte_reader_get_uint32_be (&reader, &img_description_len)) - goto error; - if (!gst_byte_reader_skip (&reader, img_description_len)) - goto error; - - if (!gst_byte_reader_skip (&reader, 4 * 4)) - goto error; - - if (!gst_byte_reader_get_uint32_be (&reader, &img_len)) - goto error; - - if (!flacparse->tags) - flacparse->tags = gst_tag_list_new (); - - gst_tag_list_add_id3_image (flacparse->tags, - data + gst_byte_reader_get_pos (&reader), img_len, img_type); - - if (gst_tag_list_is_empty (flacparse->tags)) { - gst_tag_list_free (flacparse->tags); - flacparse->tags = NULL; - } - - return TRUE; - -error: - GST_ERROR_OBJECT (flacparse, "Error reading data"); - return FALSE; -} - -static gboolean -gst_flac_parse_handle_seektable (GstFlacParse * flacparse, GstBuffer * buffer) -{ - - GST_DEBUG_OBJECT (flacparse, "storing seektable"); - /* only store for now; - * offset of the first frame is needed to get real info */ - flacparse->seektable = gst_buffer_ref (buffer); - - return TRUE; -} - -static void -gst_flac_parse_process_seektable (GstFlacParse * flacparse, gint64 boffset) -{ - GstByteReader br; - gint64 offset = 0, samples = 0; - - GST_DEBUG_OBJECT (flacparse, - "parsing seektable; base offset %" G_GINT64_FORMAT, boffset); - - if (boffset <= 0) - goto done; - - gst_byte_reader_init_from_buffer (&br, flacparse->seektable); - /* skip header */ - if (!gst_byte_reader_skip (&br, 4)) - goto done; - - /* seekpoints */ - while (gst_byte_reader_get_remaining (&br)) { - if (!gst_byte_reader_get_int64_be (&br, &samples)) - break; - if (!gst_byte_reader_get_int64_be (&br, &offset)) - break; - if (!gst_byte_reader_skip (&br, 2)) - break; - - GST_LOG_OBJECT (flacparse, "samples %" G_GINT64_FORMAT " -> offset %" - G_GINT64_FORMAT, samples, offset); - - /* sanity check */ - if (G_LIKELY (offset > 0 && samples > 0)) { - gst_base_parse_add_index_entry (GST_BASE_PARSE (flacparse), - boffset + offset, gst_util_uint64_scale (samples, GST_SECOND, - flacparse->samplerate), TRUE, FALSE); - } - } - -done: - gst_buffer_unref (flacparse->seektable); - flacparse->seektable = NULL; -} - -static void -_value_array_append_buffer (GValue * array_val, GstBuffer * buf) -{ - GValue value = { 0, }; - - g_value_init (&value, GST_TYPE_BUFFER); - /* copy buffer to avoid problems with circular refcounts */ - buf = gst_buffer_copy (buf); - /* again, for good measure */ - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); - gst_value_set_buffer (&value, buf); - gst_buffer_unref (buf); - gst_value_array_append_value (array_val, &value); - g_value_unset (&value); -} - -static gboolean -gst_flac_parse_handle_headers (GstFlacParse * flacparse) -{ - GstBuffer *vorbiscomment = NULL; - GstBuffer *streaminfo = NULL; - GstBuffer *marker = NULL; - GValue array = { 0, }; - GstCaps *caps; - GList *l; - gboolean res = TRUE; - - caps = gst_caps_new_simple ("audio/x-flac", - "channels", G_TYPE_INT, flacparse->channels, - "framed", G_TYPE_BOOLEAN, TRUE, - "rate", G_TYPE_INT, flacparse->samplerate, NULL); - - if (!flacparse->headers) - goto push_headers; - - for (l = flacparse->headers; l; l = l->next) { - GstBuffer *header = l->data; - const guint8 *data = GST_BUFFER_DATA (header); - guint size = GST_BUFFER_SIZE (header); - - GST_BUFFER_FLAG_SET (header, GST_BUFFER_FLAG_IN_CAPS); - - if (size == 4 && memcmp (data, "fLaC", 4) == 0) { - marker = header; - } else if (size > 1 && (data[0] & 0x7f) == 0) { - streaminfo = header; - } else if (size > 1 && (data[0] & 0x7f) == 4) { - vorbiscomment = header; - } - } - - if (marker == NULL || streaminfo == NULL || vorbiscomment == NULL) { - GST_WARNING_OBJECT (flacparse, - "missing header %p %p %p, muxing into container " - "formats may be broken", marker, streaminfo, vorbiscomment); - goto push_headers; - } - - g_value_init (&array, GST_TYPE_ARRAY); - - /* add marker including STREAMINFO header */ - { - GstBuffer *buf; - guint16 num; - - /* minus one for the marker that is merged with streaminfo here */ - num = g_list_length (flacparse->headers) - 1; - - buf = gst_buffer_new_and_alloc (13 + GST_BUFFER_SIZE (streaminfo)); - GST_BUFFER_DATA (buf)[0] = 0x7f; - memcpy (GST_BUFFER_DATA (buf) + 1, "FLAC", 4); - GST_BUFFER_DATA (buf)[5] = 0x01; /* mapping version major */ - GST_BUFFER_DATA (buf)[6] = 0x00; /* mapping version minor */ - GST_BUFFER_DATA (buf)[7] = (num & 0xFF00) >> 8; - GST_BUFFER_DATA (buf)[8] = (num & 0x00FF) >> 0; - memcpy (GST_BUFFER_DATA (buf) + 9, "fLaC", 4); - memcpy (GST_BUFFER_DATA (buf) + 13, GST_BUFFER_DATA (streaminfo), - GST_BUFFER_SIZE (streaminfo)); - _value_array_append_buffer (&array, buf); - gst_buffer_unref (buf); - } - - /* add VORBISCOMMENT header */ - _value_array_append_buffer (&array, vorbiscomment); - - /* add other headers, if there are any */ - for (l = flacparse->headers; l; l = l->next) { - if (GST_BUFFER_CAST (l->data) != marker && - GST_BUFFER_CAST (l->data) != streaminfo && - GST_BUFFER_CAST (l->data) != vorbiscomment) { - _value_array_append_buffer (&array, GST_BUFFER_CAST (l->data)); - } - } - - gst_structure_set_value (gst_caps_get_structure (caps, 0), - "streamheader", &array); - g_value_unset (&array); - -push_headers: - - gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (GST_BASE_PARSE (flacparse)), caps); - gst_caps_unref (caps); - - /* push header buffers; update caps, so when we push the first buffer the - * negotiated caps will change to caps that include the streamheader field */ - while (flacparse->headers) { - GstBuffer *buf = GST_BUFFER (flacparse->headers->data); - GstFlowReturn ret; - GstBaseParseFrame frame; - - flacparse->headers = - g_list_delete_link (flacparse->headers, flacparse->headers); - buf = gst_buffer_make_metadata_writable (buf); - gst_buffer_set_caps (buf, - GST_PAD_CAPS (GST_BASE_PARSE_SRC_PAD (GST_BASE_PARSE (flacparse)))); - - /* init, set and give away frame */ - gst_base_parse_frame_init (GST_BASE_PARSE (flacparse), &frame); - frame.buffer = buf; - frame.overhead = -1; - ret = gst_base_parse_push_frame (GST_BASE_PARSE (flacparse), &frame); - if (ret != GST_FLOW_OK) { - res = FALSE; - break; - } - } - g_list_foreach (flacparse->headers, (GFunc) gst_mini_object_unref, NULL); - g_list_free (flacparse->headers); - flacparse->headers = NULL; - - return res; -} - -static gboolean -gst_flac_parse_generate_headers (GstFlacParse * flacparse) -{ - GstBuffer *marker, *streaminfo, *vorbiscomment; - guint8 *data; - - marker = gst_buffer_new_and_alloc (4); - memcpy (GST_BUFFER_DATA (marker), "fLaC", 4); - GST_BUFFER_TIMESTAMP (marker) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (marker) = GST_CLOCK_TIME_NONE; - GST_BUFFER_OFFSET (marker) = 0; - GST_BUFFER_OFFSET_END (marker) = 0; - flacparse->headers = g_list_append (flacparse->headers, marker); - - streaminfo = gst_buffer_new_and_alloc (4 + 34); - data = GST_BUFFER_DATA (streaminfo); - memset (data, 0, 4 + 34); - - /* metadata block header */ - data[0] = 0x00; /* is_last = 0; type = 0; */ - data[1] = 0x00; /* length = 34; */ - data[2] = 0x00; - data[3] = 0x22; - - /* streaminfo */ - - data[4] = (flacparse->block_size >> 8) & 0xff; /* min blocksize = blocksize; */ - data[5] = (flacparse->block_size) & 0xff; - data[6] = (flacparse->block_size >> 8) & 0xff; /* max blocksize = blocksize; */ - data[7] = (flacparse->block_size) & 0xff; - - data[8] = 0x00; /* min framesize = 0; */ - data[9] = 0x00; - data[10] = 0x00; - data[11] = 0x00; /* max framesize = 0; */ - data[12] = 0x00; - data[13] = 0x00; - - data[14] = (flacparse->samplerate >> 12) & 0xff; - data[15] = (flacparse->samplerate >> 4) & 0xff; - data[16] = (flacparse->samplerate >> 0) & 0xf0; - - data[16] |= (flacparse->channels - 1) << 1; - - data[16] |= ((flacparse->bps - 1) >> 4) & 0x01; - data[17] = (((flacparse->bps - 1)) & 0x0f) << 4; - - { - gint64 duration; - GstFormat fmt = GST_FORMAT_TIME; - - if (gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE - (flacparse)), &fmt, &duration) && fmt == GST_FORMAT_TIME) { - duration = GST_CLOCK_TIME_TO_FRAMES (duration, flacparse->samplerate); - - data[17] |= (duration >> 32) & 0xff; - data[18] |= (duration >> 24) & 0xff; - data[19] |= (duration >> 16) & 0xff; - data[20] |= (duration >> 8) & 0xff; - data[21] |= (duration >> 0) & 0xff; - } - } - /* MD5 = 0; */ - - GST_BUFFER_TIMESTAMP (streaminfo) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (streaminfo) = GST_CLOCK_TIME_NONE; - GST_BUFFER_OFFSET (streaminfo) = 0; - GST_BUFFER_OFFSET_END (streaminfo) = 0; - flacparse->headers = g_list_append (flacparse->headers, streaminfo); - - /* empty vorbiscomment */ - { - GstTagList *taglist = gst_tag_list_new (); - guchar header[4]; - guint size; - - header[0] = 0x84; /* is_last = 1; type = 4; */ - - vorbiscomment = - gst_tag_list_to_vorbiscomment_buffer (taglist, header, - sizeof (header), NULL); - gst_tag_list_free (taglist); - - /* Get rid of framing bit */ - if (GST_BUFFER_DATA (vorbiscomment)[GST_BUFFER_SIZE (vorbiscomment) - - 1] == 1) { - GstBuffer *sub; - - sub = - gst_buffer_create_sub (vorbiscomment, 0, - GST_BUFFER_SIZE (vorbiscomment) - 1); - gst_buffer_unref (vorbiscomment); - vorbiscomment = sub; - } - - size = GST_BUFFER_SIZE (vorbiscomment) - 4; - GST_BUFFER_DATA (vorbiscomment)[1] = ((size & 0xFF0000) >> 16); - GST_BUFFER_DATA (vorbiscomment)[2] = ((size & 0x00FF00) >> 8); - GST_BUFFER_DATA (vorbiscomment)[3] = (size & 0x0000FF); - - GST_BUFFER_TIMESTAMP (vorbiscomment) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (vorbiscomment) = GST_CLOCK_TIME_NONE; - GST_BUFFER_OFFSET (vorbiscomment) = 0; - GST_BUFFER_OFFSET_END (vorbiscomment) = 0; - flacparse->headers = g_list_append (flacparse->headers, vorbiscomment); - } - - return TRUE; -} - -static GstFlowReturn -gst_flac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (parse); - GstBuffer *buffer = frame->buffer; - const guint8 *data = GST_BUFFER_DATA (buffer); - - if (flacparse->state == GST_FLAC_PARSE_STATE_INIT) { - GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; - GST_BUFFER_OFFSET (buffer) = 0; - GST_BUFFER_OFFSET_END (buffer) = 0; - - /* 32 bits metadata block */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), 4); - flacparse->state = GST_FLAC_PARSE_STATE_HEADERS; - - flacparse->headers = - g_list_append (flacparse->headers, gst_buffer_ref (buffer)); - - return GST_BASE_PARSE_FLOW_DROPPED; - } else if (flacparse->state == GST_FLAC_PARSE_STATE_HEADERS) { - gboolean is_last = ((data[0] & 0x80) == 0x80); - guint type = (data[0] & 0x7F); - - if (type == 127) { - GST_WARNING_OBJECT (flacparse, "Invalid metadata block type"); - return GST_BASE_PARSE_FLOW_DROPPED; - } - - GST_DEBUG_OBJECT (flacparse, "Handling metadata block of type %u", type); - - switch (type) { - case 0: /* STREAMINFO */ - if (!gst_flac_parse_handle_streaminfo (flacparse, buffer)) - return GST_FLOW_ERROR; - break; - case 3: /* SEEKTABLE */ - if (!gst_flac_parse_handle_seektable (flacparse, buffer)) - return GST_FLOW_ERROR; - break; - case 4: /* VORBIS_COMMENT */ - if (!gst_flac_parse_handle_vorbiscomment (flacparse, buffer)) - return GST_FLOW_ERROR; - break; - case 6: /* PICTURE */ - if (!gst_flac_parse_handle_picture (flacparse, buffer)) - return GST_FLOW_ERROR; - break; - case 1: /* PADDING */ - case 2: /* APPLICATION */ - case 5: /* CUESHEET */ - default: /* RESERVED */ - break; - } - - GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; - GST_BUFFER_OFFSET (buffer) = 0; - GST_BUFFER_OFFSET_END (buffer) = 0; - - flacparse->headers = - g_list_append (flacparse->headers, gst_buffer_ref (buffer)); - - if (is_last) { - if (!gst_flac_parse_handle_headers (flacparse)) - return GST_FLOW_ERROR; - - /* Minimal size of a frame header */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), MAX (9, - flacparse->min_framesize)); - flacparse->state = GST_FLAC_PARSE_STATE_DATA; - } - - /* DROPPED because we pushed already or will push all headers manually */ - return GST_BASE_PARSE_FLOW_DROPPED; - } else { - if (flacparse->offset != GST_BUFFER_OFFSET (buffer)) { - FrameHeaderCheckReturn ret; - - flacparse->offset = GST_BUFFER_OFFSET (buffer); - ret = - gst_flac_parse_frame_header_is_valid (flacparse, - GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), TRUE, NULL); - if (ret != FRAME_HEADER_VALID) { - GST_ERROR_OBJECT (flacparse, - "Baseclass didn't provide a complete frame"); - return GST_FLOW_ERROR; - } - } - - if (flacparse->block_size == 0) { - GST_ERROR_OBJECT (flacparse, "Unparsed frame"); - return GST_FLOW_ERROR; - } - - if (flacparse->seektable) - gst_flac_parse_process_seektable (flacparse, GST_BUFFER_OFFSET (buffer)); - - if (flacparse->state == GST_FLAC_PARSE_STATE_GENERATE_HEADERS) { - if (flacparse->blocking_strategy == 1) { - GST_WARNING_OBJECT (flacparse, - "Generating headers for variable blocksize streams not supported"); - - if (!gst_flac_parse_handle_headers (flacparse)) - return GST_FLOW_ERROR; - } else { - GST_DEBUG_OBJECT (flacparse, "Generating headers"); - - if (!gst_flac_parse_generate_headers (flacparse)) - return GST_FLOW_ERROR; - - if (!gst_flac_parse_handle_headers (flacparse)) - return GST_FLOW_ERROR; - } - flacparse->state = GST_FLAC_PARSE_STATE_DATA; - } - - /* also cater for oggmux metadata */ - if (flacparse->blocking_strategy == 0) { - GST_BUFFER_TIMESTAMP (buffer) = - gst_util_uint64_scale (flacparse->sample_number, - flacparse->block_size * GST_SECOND, flacparse->samplerate); - GST_BUFFER_OFFSET_END (buffer) = - flacparse->sample_number * flacparse->block_size + - flacparse->block_size; - } else { - GST_BUFFER_TIMESTAMP (buffer) = - gst_util_uint64_scale (flacparse->sample_number, GST_SECOND, - flacparse->samplerate); - GST_BUFFER_OFFSET_END (buffer) = - flacparse->sample_number + flacparse->block_size; - } - GST_BUFFER_OFFSET (buffer) = - gst_util_uint64_scale (GST_BUFFER_OFFSET_END (buffer), GST_SECOND, - flacparse->samplerate); - GST_BUFFER_DURATION (buffer) = - GST_BUFFER_OFFSET (buffer) - GST_BUFFER_TIMESTAMP (buffer); - - /* To simplify, we just assume that it's a fixed size header and ignore - * subframe headers. The first could lead us to being off by 88 bits and - * the second even less, so the total inaccuracy is negligible. */ - frame->overhead = 7; - - /* Minimal size of a frame header */ - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (flacparse), MAX (9, - flacparse->min_framesize)); - - flacparse->offset = -1; - flacparse->blocking_strategy = 0; - flacparse->block_size = 0; - flacparse->sample_number = 0; - return GST_FLOW_OK; - } -} - -static GstFlowReturn -gst_flac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstFlacParse *flacparse = GST_FLAC_PARSE (parse); - - /* Push tags */ - if (flacparse->tags) { - gst_element_found_tags (GST_ELEMENT (flacparse), flacparse->tags); - flacparse->tags = NULL; - } - - frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP; - - return GST_FLOW_OK; -} diff --git a/gst/audioparsers/gstflacparse.h b/gst/audioparsers/gstflacparse.h deleted file mode 100644 index 664b2a6..0000000 --- a/gst/audioparsers/gstflacparse.h +++ /dev/null @@ -1,92 +0,0 @@ -/* GStreamer - * - * Copyright (C) 2008 Sebastian Dröge . - * Copyright (C) 2009 Mark Nauwelaerts - * Copyright (C) 2009 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_FLAC_PARSE_H__ -#define __GST_FLAC_PARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_FLAC_PARSE (gst_flac_parse_get_type()) -#define GST_FLAC_PARSE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_FLAC_PARSE,GstFlacParse)) -#define GST_FLAC_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_FLAC_PARSE,GstFlacParseClass)) -#define GST_FLAC_PARSE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_FLAC_PARSE,GstFlacParseClass)) -#define GST_IS_FLAC_PARSE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_FLAC_PARSE)) -#define GST_IS_FLAC_PARSE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_FLAC_PARSE)) -#define GST_FLAC_PARSE_CAST(obj) ((GstFlacParse *)(obj)) - -typedef struct _GstFlacParse GstFlacParse; -typedef struct _GstFlacParseClass GstFlacParseClass; - -typedef enum { - GST_FLAC_PARSE_STATE_INIT, - GST_FLAC_PARSE_STATE_HEADERS, - GST_FLAC_PARSE_STATE_GENERATE_HEADERS, - GST_FLAC_PARSE_STATE_DATA -} GstFlacParseState; - -typedef struct { - guint8 type; -} GstFlacParseSubFrame; - -struct _GstFlacParse { - GstBaseParse parent; - - /* Properties */ - gboolean check_frame_checksums; - - GstFlacParseState state; - - gint64 upstream_length; - - /* STREAMINFO content */ - guint16 min_blocksize, max_blocksize; - guint32 min_framesize, max_framesize; - guint32 samplerate; - guint8 channels; - guint8 bps; - guint64 total_samples; - - /* Current frame */ - guint64 offset; - guint8 blocking_strategy; - guint16 block_size; - guint64 sample_number; - - GstTagList *tags; - - GList *headers; - GstBuffer *seektable; -}; - -struct _GstFlacParseClass { - GstBaseParseClass parent_class; -}; - -GType gst_flac_parse_get_type (void); - -G_END_DECLS - -#endif /* __GST_FLAC_PARSE_H__ */ diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c deleted file mode 100644 index a9eabdc..0000000 --- a/gst/audioparsers/gstmpegaudioparse.c +++ /dev/null @@ -1,1252 +0,0 @@ -/* GStreamer MPEG audio parser - * Copyright (C) 2006-2007 Jan Schmidt - * Copyright (C) 2010 Mark Nauwelaerts - * Copyright (C) 2010 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ -/** - * SECTION:element-mpegaudioparse - * @short_description: MPEG audio parser - * @see_also: #GstAmrParse, #GstAACParse - * - * Parses and frames mpeg1 audio streams. Provides seeking. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink - * ]| - * - */ - -/* FIXME: we should make the base class (GstBaseParse) aware of the - * XING seek table somehow, so it can use it properly for things like - * accurate seeks. Currently it can only do a lookup via the convert function, - * but then doesn't know what the result represents exactly. One could either - * add a vfunc for index lookup, or just make mpegaudioparse populate the - * base class's index via the API provided. - */ -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstmpegaudioparse.h" -#include - -GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug); -#define GST_CAT_DEFAULT mpeg_audio_parse_debug - -#define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1 -#define MPEG_AUDIO_CHANNEL_MODE_STEREO 0 -#define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1 -#define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2 -#define MPEG_AUDIO_CHANNEL_MODE_MONO 3 - -#define CRC_UNKNOWN -1 -#define CRC_PROTECTED 0 -#define CRC_NOT_PROTECTED 1 - -#define XING_FRAMES_FLAG 0x0001 -#define XING_BYTES_FLAG 0x0002 -#define XING_TOC_FLAG 0x0004 -#define XING_VBR_SCALE_FLAG 0x0008 - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " - "mpegversion = (int) 1, " - "layer = (int) [ 1, 3 ], " - "rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ]," - "parsed=(boolean) true") - ); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false") - ); - -static void gst_mpeg_audio_parse_finalize (GObject * object); - -static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse); -static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse); -static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * size, gint * skipsize); -static GstFlowReturn gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); -static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); -static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse, - GstFormat src_format, gint64 src_value, - GstFormat dest_format, gint64 * dest_value); - -GST_BOILERPLATE (GstMpegAudioParse, gst_mpeg_audio_parse, GstBaseParse, - GST_TYPE_BASE_PARSE); - -#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE \ - (gst_mpeg_audio_channel_mode_get_type()) - -static const GEnumValue mpeg_audio_channel_mode[] = { - {MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"}, - {MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"}, - {MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"}, - {MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"}, - {MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"}, - {0, NULL, NULL}, -}; - -static GType -gst_mpeg_audio_channel_mode_get_type (void) -{ - static GType mpeg_audio_channel_mode_type = 0; - - if (!mpeg_audio_channel_mode_type) { - mpeg_audio_channel_mode_type = - g_enum_register_static ("GstMpegAudioChannelMode", - mpeg_audio_channel_mode); - } - return mpeg_audio_channel_mode_type; -} - -static const gchar * -gst_mpeg_audio_channel_mode_get_nick (gint mode) -{ - guint i; - for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) { - if (mpeg_audio_channel_mode[i].value == mode) - return mpeg_audio_channel_mode[i].value_nick; - } - return NULL; -} - -static void -gst_mpeg_audio_parse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - - gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser", - "Codec/Parser/Audio", - "Parses and frames mpeg1 audio streams (levels 1-3), provides seek", - "Jan Schmidt ," - "Mark Nauwelaerts "); -} - -static void -gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass) -{ - GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); - GObjectClass *object_class = G_OBJECT_CLASS (klass); - - GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0, - "MPEG1 audio stream parser"); - - object_class->finalize = gst_mpeg_audio_parse_finalize; - - parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start); - parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_check_valid_frame); - parse_class->parse_frame = - GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_parse_frame); - parse_class->pre_push_frame = - GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame); - parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert); - - /* register tags */ -#define GST_TAG_CRC "has-crc" -#define GST_TAG_MODE "channel-mode" - - gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN, - "has crc", "Using CRC", NULL); - gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING, - "channel mode", "MPEG audio channel mode", NULL); - - g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE); -} - -static void -gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse) -{ - mp3parse->channels = -1; - mp3parse->rate = -1; - mp3parse->sent_codec_tag = FALSE; - mp3parse->last_posted_crc = CRC_UNKNOWN; - mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN; - - mp3parse->hdr_bitrate = 0; - - mp3parse->xing_flags = 0; - mp3parse->xing_bitrate = 0; - mp3parse->xing_frames = 0; - mp3parse->xing_total_time = 0; - mp3parse->xing_bytes = 0; - mp3parse->xing_vbr_scale = 0; - memset (mp3parse->xing_seek_table, 0, 100); - memset (mp3parse->xing_seek_table_inverse, 0, 256); - - mp3parse->vbri_bitrate = 0; - mp3parse->vbri_frames = 0; - mp3parse->vbri_total_time = 0; - mp3parse->vbri_bytes = 0; - mp3parse->vbri_seek_points = 0; - g_free (mp3parse->vbri_seek_table); - mp3parse->vbri_seek_table = NULL; - - mp3parse->encoder_delay = 0; - mp3parse->encoder_padding = 0; -} - -static void -gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse, - GstMpegAudioParseClass * klass) -{ - gst_mpeg_audio_parse_reset (mp3parse); -} - -static void -gst_mpeg_audio_parse_finalize (GObject * object) -{ - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_mpeg_audio_parse_start (GstBaseParse * parse) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - - gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), 1024); - GST_DEBUG_OBJECT (parse, "starting"); - - gst_mpeg_audio_parse_reset (mp3parse); - - return TRUE; -} - -static gboolean -gst_mpeg_audio_parse_stop (GstBaseParse * parse) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - - GST_DEBUG_OBJECT (parse, "stopping"); - - gst_mpeg_audio_parse_reset (mp3parse); - - return TRUE; -} - -static const guint mp3types_bitrates[2][3][16] = { - { - {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,}, - {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,}, - {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,} - }, - { - {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,} - }, -}; - -static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, -{22050, 24000, 16000}, -{11025, 12000, 8000} -}; - -static inline guint -mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header, - guint * put_version, guint * put_layer, guint * put_channels, - guint * put_bitrate, guint * put_samplerate, guint * put_mode, - guint * put_crc) -{ - guint length; - gulong mode, samplerate, bitrate, layer, channels, padding, crc; - gulong version; - gint lsf, mpg25; - - if (header & (1 << 20)) { - lsf = (header & (1 << 19)) ? 0 : 1; - mpg25 = 0; - } else { - lsf = 1; - mpg25 = 1; - } - - version = 1 + lsf + mpg25; - - layer = 4 - ((header >> 17) & 0x3); - - crc = (header >> 16) & 0x1; - - bitrate = (header >> 12) & 0xF; - bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; - /* The caller has ensured we have a valid header, so bitrate can't be - zero here. */ - g_assert (bitrate != 0); - - samplerate = (header >> 10) & 0x3; - samplerate = mp3types_freqs[lsf + mpg25][samplerate]; - - padding = (header >> 9) & 0x1; - - mode = (header >> 6) & 0x3; - channels = (mode == 3) ? 1 : 2; - - switch (layer) { - case 1: - length = 4 * ((bitrate * 12) / samplerate + padding); - break; - case 2: - length = (bitrate * 144) / samplerate + padding; - break; - default: - case 3: - length = (bitrate * 144) / (samplerate << lsf) + padding; - break; - } - - GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes", - length); - GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, " - "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version, - layer, channels, gst_mpeg_audio_channel_mode_get_nick (mode)); - - if (put_version) - *put_version = version; - if (put_layer) - *put_layer = layer; - if (put_channels) - *put_channels = channels; - if (put_bitrate) - *put_bitrate = bitrate; - if (put_samplerate) - *put_samplerate = samplerate; - if (put_mode) - *put_mode = mode; - if (put_crc) - *put_crc = crc; - - return length; -} - -/* Minimum number of consecutive, valid-looking frames to consider - * for resyncing */ -#define MIN_RESYNC_FRAMES 3 - -/* Perform extended validation to check that subsequent headers match - * the first header given here in important characteristics, to avoid - * false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive - * frames to match their major characteristics. - * - * If at_eos is set to TRUE, we just check that we don't find any invalid - * frames in whatever data is available, rather than requiring a full - * MIN_RESYNC_FRAMES of data. - * - * Returns TRUE if we've seen enough data to validate or reject the frame. - * If TRUE is returned, then *valid contains TRUE if it validated, or false - * if we decided it was false sync. - * If FALSE is returned, then *valid contains minimum needed data. - */ -static gboolean -gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf, - guint32 header, int bpf, gboolean at_eos, gint * valid) -{ - guint32 next_header; - const guint8 *data; - guint available; - int frames_found = 1; - int offset = bpf; - - available = GST_BUFFER_SIZE (buf); - data = GST_BUFFER_DATA (buf); - - while (frames_found < MIN_RESYNC_FRAMES) { - /* Check if we have enough data for all these frames, plus the next - frame header. */ - if (available < offset + 4) { - if (at_eos) { - /* Running out of data at EOS is fine; just accept it */ - *valid = TRUE; - return TRUE; - } else { - *valid = offset + 4; - return FALSE; - } - } - - next_header = GST_READ_UINT32_BE (data + offset); - GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d", - offset, (unsigned int) header, (unsigned int) next_header, bpf); - -/* mask the bits which are allowed to differ between frames */ -#define HDRMASK ~((0xF << 12) /* bitrate */ | \ - (0x1 << 9) /* padding */ | \ - (0xf << 4) /* mode|mode extension */ | \ - (0xf)) /* copyright|emphasis */ - - if ((next_header & HDRMASK) != (header & HDRMASK)) { - /* If any of the unmasked bits don't match, then it's not valid */ - GST_DEBUG_OBJECT (mp3parse, "next header doesn't match " - "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)", - (guint) header, (guint) header & HDRMASK, (guint) next_header, - (guint) next_header & HDRMASK, bpf); - *valid = FALSE; - return TRUE; - } else if ((((next_header >> 12) & 0xf) == 0) || - (((next_header >> 12) & 0xf) == 0xf)) { - /* The essential parts were the same, but the bitrate held an - invalid value - also reject */ - GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)"); - *valid = FALSE; - return TRUE; - } - - bpf = mp3_type_frame_length_from_header (mp3parse, next_header, - NULL, NULL, NULL, NULL, NULL, NULL, NULL); - - offset += bpf; - frames_found++; - } - - *valid = TRUE; - return TRUE; -} - -static gboolean -gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse, - unsigned long head) -{ - GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head); - /* if it's not a valid sync */ - if ((head & 0xffe00000) != 0xffe00000) { - GST_WARNING_OBJECT (mp3parse, "invalid sync"); - return FALSE; - } - /* if it's an invalid MPEG version */ - if (((head >> 19) & 3) == 0x1) { - GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx", - (head >> 19) & 3); - return FALSE; - } - /* if it's an invalid layer */ - if (!((head >> 17) & 3)) { - GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3); - return FALSE; - } - /* if it's an invalid bitrate */ - if (((head >> 12) & 0xf) == 0x0) { - GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx." - "Free format files are not supported yet", (head >> 12) & 0xf); - return FALSE; - } - if (((head >> 12) & 0xf) == 0xf) { - GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf); - return FALSE; - } - /* if it's an invalid samplerate */ - if (((head >> 10) & 0x3) == 0x3) { - GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx", - (head >> 10) & 0x3); - return FALSE; - } - - if ((head & 0x3) == 0x2) { - /* Ignore this as there are some files with emphasis 0x2 that can - * be played fine. See BGO #537235 */ - GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3); - } - - return TRUE; -} - -static gboolean -gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - GstBuffer *buf = frame->buffer; - GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - gint off, bpf; - gboolean sync, drain, valid, caps_change; - guint32 header; - guint bitrate, layer, rate, channels, version, mode, crc; - - if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6)) - return FALSE; - - off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000, - 0, GST_BUFFER_SIZE (buf)); - - GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); - - /* didn't find anything that looks like a sync word, skip */ - if (off < 0) { - *skipsize = GST_BUFFER_SIZE (buf) - 3; - return FALSE; - } - - /* possible frame header, but not at offset 0? skip bytes before sync */ - if (off > 0) { - *skipsize = off; - return FALSE; - } - - /* make sure the values in the frame header look sane */ - header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)); - if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) { - *skipsize = 1; - return FALSE; - } - - GST_LOG_OBJECT (parse, "got frame"); - - bpf = mp3_type_frame_length_from_header (mp3parse, header, - &version, &layer, &channels, &bitrate, &rate, &mode, &crc); - g_assert (bpf != 0); - - if (channels != mp3parse->channels || rate != mp3parse->rate || - layer != mp3parse->layer || version != mp3parse->version) - caps_change = TRUE; - else - caps_change = FALSE; - - sync = GST_BASE_PARSE_FRAME_SYNC (frame); - drain = GST_BASE_PARSE_FRAME_DRAIN (frame); - - if (!drain && (!sync || caps_change)) { - if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, drain, - &valid)) { - /* not enough data */ - gst_base_parse_set_min_frame_size (parse, valid); - *skipsize = 0; - return FALSE; - } else { - if (!valid) { - *skipsize = off + 2; - return FALSE; - } - } - } else if (drain && !sync && caps_change && mp3parse->rate > 0) { - /* avoid caps jitter that we can't be sure of */ - *skipsize = off + 2; - return FALSE; - } - - *framesize = bpf; - return TRUE; -} - -static void -gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse, - GstBuffer * buf) -{ - const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */ - const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */ - const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */ - const guint32 lame_id = 0x4c414d45; /* 'LAME' in hex */ - gint offset; - guint64 avail; - gint64 upstream_total_bytes = 0; - GstFormat fmt = GST_FORMAT_BYTES; - guint32 read_id; - const guint8 *data; - GstBaseParseSeekable seekable; - guint bitrate; - - if (mp3parse->sent_codec_tag) - return; - - /* Check first frame for Xing info */ - if (mp3parse->version == 1) { /* MPEG-1 file */ - if (mp3parse->channels == 1) - offset = 0x11; - else - offset = 0x20; - } else { /* MPEG-2 header */ - if (mp3parse->channels == 1) - offset = 0x09; - else - offset = 0x11; - } - /* Skip the 4 bytes of the MP3 header too */ - offset += 4; - - /* Check if we have enough data to read the Xing header */ - avail = GST_BUFFER_SIZE (buf); - data = GST_BUFFER_DATA (buf); - if (avail < offset + 8) - return; - - /* The header starts at the provided offset */ - data += offset; - - /* obtain real upstream total bytes */ - fmt = GST_FORMAT_BYTES; - if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE - (mp3parse)), &fmt, &upstream_total_bytes)) - upstream_total_bytes = 0; - - read_id = GST_READ_UINT32_BE (data); - if (read_id == xing_id || read_id == info_id) { - guint32 xing_flags; - guint bytes_needed = offset + 8; - gint64 total_bytes; - GstClockTime total_time; - - GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id); - - /* Read 4 base bytes of flags, big-endian */ - xing_flags = GST_READ_UINT32_BE (data + 4); - if (xing_flags & XING_FRAMES_FLAG) - bytes_needed += 4; - if (xing_flags & XING_BYTES_FLAG) - bytes_needed += 4; - if (xing_flags & XING_TOC_FLAG) - bytes_needed += 100; - if (xing_flags & XING_VBR_SCALE_FLAG) - bytes_needed += 4; - if (avail < bytes_needed) { - GST_DEBUG_OBJECT (mp3parse, - "Not enough data to read Xing header (need %d)", bytes_needed); - return; - } - - GST_DEBUG_OBJECT (mp3parse, "Reading Xing header"); - mp3parse->xing_flags = xing_flags; - - data = GST_BUFFER_DATA (buf); - data += offset + 8; - - if (xing_flags & XING_FRAMES_FLAG) { - mp3parse->xing_frames = GST_READ_UINT32_BE (data); - if (mp3parse->xing_frames == 0) { - GST_WARNING_OBJECT (mp3parse, - "Invalid number of frames in Xing header"); - mp3parse->xing_flags &= ~XING_FRAMES_FLAG; - } else { - mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND, - (guint64) (mp3parse->xing_frames) * (mp3parse->spf), - mp3parse->rate); - } - - data += 4; - } else { - mp3parse->xing_frames = 0; - mp3parse->xing_total_time = 0; - } - - if (xing_flags & XING_BYTES_FLAG) { - mp3parse->xing_bytes = GST_READ_UINT32_BE (data); - if (mp3parse->xing_bytes == 0) { - GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header"); - mp3parse->xing_flags &= ~XING_BYTES_FLAG; - } - data += 4; - } else { - mp3parse->xing_bytes = 0; - } - - /* If we know the upstream size and duration, compute the - * total bitrate, rounded up to the nearest kbit/sec */ - if ((total_time = mp3parse->xing_total_time) && - (total_bytes = mp3parse->xing_bytes)) { - mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes, - 8 * GST_SECOND, total_time); - mp3parse->xing_bitrate += 500; - mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000; - } - - if (xing_flags & XING_TOC_FLAG) { - int i, percent = 0; - guchar *table = mp3parse->xing_seek_table; - guchar old = 0, new; - guint first; - - first = data[0]; - GST_DEBUG_OBJECT (mp3parse, - "Subtracting initial offset of %d bytes from Xing TOC", first); - - /* xing seek table: percent time -> 1/256 bytepos */ - for (i = 0; i < 100; i++) { - new = data[i] - first; - if (old > new) { - GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC"); - mp3parse->xing_flags &= ~XING_TOC_FLAG; - goto skip_toc; - } - mp3parse->xing_seek_table[i] = old = new; - } - - /* build inverse table: 1/256 bytepos -> 1/100 percent time */ - for (i = 0; i < 256; i++) { - while (percent < 99 && table[percent + 1] <= i) - percent++; - - if (table[percent] == i) { - mp3parse->xing_seek_table_inverse[i] = percent * 100; - } else if (table[percent] < i && percent < 99) { - gdouble fa, fb, fx; - gint a = percent, b = percent + 1; - - fa = table[a]; - fb = table[b]; - fx = (b - a) / (fb - fa) * (i - fa) + a; - mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); - } else if (percent == 99) { - gdouble fa, fb, fx; - gint a = percent, b = 100; - - fa = table[a]; - fb = 256.0; - fx = (b - a) / (fb - fa) * (i - fa) + a; - mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); - } - } - skip_toc: - data += 100; - } else { - memset (mp3parse->xing_seek_table, 0, 100); - memset (mp3parse->xing_seek_table_inverse, 0, 256); - } - - if (xing_flags & XING_VBR_SCALE_FLAG) { - mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data); - data += 4; - } else - mp3parse->xing_vbr_scale = 0; - - GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %" - GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames, - GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes, - mp3parse->xing_vbr_scale); - - /* check for truncated file */ - if (upstream_total_bytes && mp3parse->xing_bytes && - mp3parse->xing_bytes * 0.8 > upstream_total_bytes) { - GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; " - "invalidating Xing header duration and size"); - mp3parse->xing_flags &= ~XING_BYTES_FLAG; - mp3parse->xing_flags &= ~XING_FRAMES_FLAG; - } - - /* Optional LAME tag? */ - if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) { - gchar lame_version[10] = { 0, }; - guint tag_rev; - guint32 encoder_delay, encoder_padding; - - memcpy (lame_version, data, 9); - data += 9; - tag_rev = data[0] >> 4; - GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'", - tag_rev, lame_version); - - /* Skip all the information we're not interested in */ - data += 12; - /* Encoder delay and end padding */ - encoder_delay = GST_READ_UINT24_BE (data); - encoder_delay >>= 12; - encoder_padding = GST_READ_UINT24_BE (data); - encoder_padding &= 0x000fff; - - mp3parse->encoder_delay = encoder_delay; - mp3parse->encoder_padding = encoder_padding; - - GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u", - encoder_delay, encoder_padding); - } - } else if (read_id == vbri_id) { - gint64 total_bytes, total_frames; - GstClockTime total_time; - guint16 nseek_points; - - GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id); - if (avail < offset + 26) { - GST_DEBUG_OBJECT (mp3parse, - "Not enough data to read VBRI header (need %d)", offset + 26); - return; - } - - GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header"); - data = GST_BUFFER_DATA (buf); - data += offset + 4; - - if (GST_READ_UINT16_BE (data) != 0x0001) { - GST_WARNING_OBJECT (mp3parse, - "Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data)); - return; - } - data += 2; - - /* Skip encoder delay */ - data += 2; - - /* Skip quality */ - data += 2; - - total_bytes = GST_READ_UINT32_BE (data); - if (total_bytes != 0) - mp3parse->vbri_bytes = total_bytes; - data += 4; - - total_frames = GST_READ_UINT32_BE (data); - if (total_frames != 0) { - mp3parse->vbri_frames = total_frames; - mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND, - (guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate); - } - data += 4; - - /* If we know the upstream size and duration, compute the - * total bitrate, rounded up to the nearest kbit/sec */ - if ((total_time = mp3parse->vbri_total_time) && - (total_bytes = mp3parse->vbri_bytes)) { - mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes, - 8 * GST_SECOND, total_time); - mp3parse->vbri_bitrate += 500; - mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000; - } - - nseek_points = GST_READ_UINT16_BE (data); - data += 2; - - if (nseek_points > 0) { - guint scale, seek_bytes, seek_frames; - gint i; - - mp3parse->vbri_seek_points = nseek_points; - - scale = GST_READ_UINT16_BE (data); - data += 2; - - seek_bytes = GST_READ_UINT16_BE (data); - data += 2; - - seek_frames = GST_READ_UINT16_BE (data); - - if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) { - GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table"); - goto out_vbri; - } - - if (avail < offset + 26 + nseek_points * seek_bytes) { - GST_WARNING_OBJECT (mp3parse, - "Not enough data to read VBRI seek table (need %d)", - offset + 26 + nseek_points * seek_bytes); - goto out_vbri; - } - - if (seek_frames * nseek_points < total_frames - seek_frames || - seek_frames * nseek_points > total_frames + seek_frames) { - GST_WARNING_OBJECT (mp3parse, - "VBRI seek table doesn't cover the complete file"); - goto out_vbri; - } - - if (avail < offset + 26) { - GST_DEBUG_OBJECT (mp3parse, - "Not enough data to read VBRI header (need %d)", - offset + 26 + nseek_points * seek_bytes); - return; - } - - data = GST_BUFFER_DATA (buf); - data += offset + 26; - - /* VBRI seek table: frame/seek_frames -> byte */ - mp3parse->vbri_seek_table = g_new (guint32, nseek_points); - if (seek_bytes == 4) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale; - data += 4; - } else if (seek_bytes == 3) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale; - data += 3; - } else if (seek_bytes == 2) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale; - data += 2; - } else /* seek_bytes == 1 */ - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale; - data += 1; - } - } - out_vbri: - - GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %" - GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames, - GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes); - - /* check for truncated file */ - if (upstream_total_bytes && mp3parse->vbri_bytes && - mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) { - GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; " - "invalidating VBRI header duration and size"); - mp3parse->vbri_valid = FALSE; - } else { - mp3parse->vbri_valid = TRUE; - } - } else { - GST_DEBUG_OBJECT (mp3parse, - "Xing, LAME or VBRI header not found in first frame"); - } - - /* set duration if tables provided a valid one */ - if (mp3parse->xing_flags & XING_FRAMES_FLAG) { - gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME, - mp3parse->xing_total_time, 0); - } - if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) { - gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME, - mp3parse->vbri_total_time, 0); - } - - /* tell baseclass how nicely we can seek, and a bitrate if one found */ - seekable = GST_BASE_PARSE_SEEK_DEFAULT; - if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes && - mp3parse->xing_total_time) - seekable = GST_BASE_PARSE_SEEK_TABLE; - - if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes && - mp3parse->vbri_total_time) - seekable = GST_BASE_PARSE_SEEK_TABLE; - - if (mp3parse->xing_bitrate) - bitrate = mp3parse->xing_bitrate; - else if (mp3parse->vbri_bitrate) - bitrate = mp3parse->vbri_bitrate; - else - bitrate = 0; - - gst_base_parse_set_seek (GST_BASE_PARSE (mp3parse), seekable, bitrate); -} - -static GstFlowReturn -gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - GstBuffer *buf = frame->buffer; - guint bitrate, layer, rate, channels, version, mode, crc; - - g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 4, GST_FLOW_ERROR); - - if (!mp3_type_frame_length_from_header (mp3parse, - GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)), - &version, &layer, &channels, &bitrate, &rate, &mode, &crc)) - goto broken_header; - - if (G_UNLIKELY (channels != mp3parse->channels || rate != mp3parse->rate || - layer != mp3parse->layer || version != mp3parse->version)) { - GstCaps *caps = gst_caps_new_simple ("audio/mpeg", - "mpegversion", G_TYPE_INT, 1, - "mpegaudioversion", G_TYPE_INT, version, - "layer", G_TYPE_INT, layer, - "rate", G_TYPE_INT, rate, - "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL); - gst_buffer_set_caps (buf, caps); - gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); - gst_caps_unref (caps); - - mp3parse->rate = rate; - mp3parse->channels = channels; - mp3parse->layer = layer; - mp3parse->version = version; - - /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */ - if (mp3parse->layer == 1) - mp3parse->spf = 384; - else if (mp3parse->layer == 2) - mp3parse->spf = 1152; - else if (mp3parse->version == 1) { - mp3parse->spf = 1152; - } else { - /* MPEG-2 or "2.5" */ - mp3parse->spf = 576; - } - - /* lead_in: - * We start pushing 9 frames earlier (29 frames for MPEG2) than - * segment start to be able to decode the first frame we want. - * 9 (29) frames are the theoretical maximum of frames that contain - * data for the current frame (bit reservoir). - * - * lead_out: - * Some mp3 streams have an offset in the timestamps, for which we have to - * push the frame *after* the end position in order for the decoder to be - * able to decode everything up until the segment.stop position. */ - gst_base_parse_set_frame_props (parse, mp3parse->rate, mp3parse->spf, - (version == 1) ? 10 : 30, 2); - } - - mp3parse->hdr_bitrate = bitrate; - - /* For first frame; check for seek tables and output a codec tag */ - gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf); - - /* store some frame info for later processing */ - mp3parse->last_crc = crc; - mp3parse->last_mode = mode; - - return GST_FLOW_OK; - -/* ERRORS */ -broken_header: - { - /* this really shouldn't ever happen */ - GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL)); - return GST_FLOW_ERROR; - } -} - -static gboolean -gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse, - GstClockTime ts, gint64 * bytepos) -{ - gint64 total_bytes; - GstClockTime total_time; - - /* If XING seek table exists use this for time->byte conversion */ - if ((mp3parse->xing_flags & XING_TOC_FLAG) && - (total_bytes = mp3parse->xing_bytes) && - (total_time = mp3parse->xing_total_time)) { - gdouble fa, fb, fx; - gdouble percent = - CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) / - gst_util_guint64_to_gdouble (total_time), 0.0, 100.0); - gint index = CLAMP (percent, 0, 99); - - fa = mp3parse->xing_seek_table[index]; - if (index < 99) - fb = mp3parse->xing_seek_table[index + 1]; - else - fb = 256.0; - - fx = fa + (fb - fa) * (percent - index); - - *bytepos = (1.0 / 256.0) * fx * total_bytes; - - return TRUE; - } - - if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) && - (total_time = mp3parse->vbri_total_time)) { - gint i, j; - gdouble a, b, fa, fb; - - i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time); - i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1); - - a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time, - mp3parse->vbri_seek_points)); - fa = 0.0; - for (j = i; j >= 0; j--) - fa += mp3parse->vbri_seek_table[j]; - - if (i + 1 < mp3parse->vbri_seek_points) { - b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time, - mp3parse->vbri_seek_points)); - fb = fa + mp3parse->vbri_seek_table[i + 1]; - } else { - b = gst_guint64_to_gdouble (total_time); - fb = total_bytes; - } - - *bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a); - - return TRUE; - } - - return FALSE; -} - -static gboolean -gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse, - gint64 bytepos, GstClockTime * ts) -{ - gint64 total_bytes; - GstClockTime total_time; - - /* If XING seek table exists use this for byte->time conversion */ - if ((mp3parse->xing_flags & XING_TOC_FLAG) && - (total_bytes = mp3parse->xing_bytes) && - (total_time = mp3parse->xing_total_time)) { - gdouble fa, fb, fx; - gdouble pos; - gint index; - - pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0); - index = CLAMP (pos, 0, 255); - fa = mp3parse->xing_seek_table_inverse[index]; - if (index < 255) - fb = mp3parse->xing_seek_table_inverse[index + 1]; - else - fb = 10000.0; - - fx = fa + (fb - fa) * (pos - index); - - *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time); - - return TRUE; - } - - if (mp3parse->vbri_seek_table && - (total_bytes = mp3parse->vbri_bytes) && - (total_time = mp3parse->vbri_total_time)) { - gint i = 0; - guint64 sum = 0; - gdouble a, b, fa, fb; - - do { - sum += mp3parse->vbri_seek_table[i]; - i++; - } while (i + 1 < mp3parse->vbri_seek_points - && sum + mp3parse->vbri_seek_table[i] < bytepos); - i--; - - a = gst_guint64_to_gdouble (sum); - fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time, - mp3parse->vbri_seek_points)); - - if (i + 1 < mp3parse->vbri_seek_points) { - b = a + mp3parse->vbri_seek_table[i + 1]; - fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time, - mp3parse->vbri_seek_points)); - } else { - b = total_bytes; - fb = gst_guint64_to_gdouble (total_time); - } - - *ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a)); - - return TRUE; - } - - return FALSE; -} - -static gboolean -gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format, - gint64 src_value, GstFormat dest_format, gint64 * dest_value) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - gboolean res = FALSE; - - if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES) - res = - gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value); - else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME) - res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value, - (GstClockTime *) dest_value); - - /* if no tables, fall back to default estimated rate based conversion */ - if (!res) - return gst_base_parse_convert_default (parse, src_format, src_value, - dest_format, dest_value); - - return res; -} - -static GstFlowReturn -gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse, - GstBaseParseFrame * frame) -{ - GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse); - GstTagList *taglist; - - /* tag sending done late enough in hook to ensure pending events - * have already been sent */ - - if (!mp3parse->sent_codec_tag) { - gchar *codec; - - /* codec tag */ - if (mp3parse->layer == 3) { - codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)", - mp3parse->version, mp3parse->layer); - } else { - codec = g_strdup_printf ("MPEG %d Audio, Layer %d", - mp3parse->version, mp3parse->layer); - } - taglist = gst_tag_list_new (); - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, - GST_TAG_AUDIO_CODEC, codec, NULL); - if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 && - mp3parse->vbri_bitrate == 0) { - /* We don't have a VBR bitrate, so post the available bitrate as - * nominal and let baseparse calculate the real bitrate */ - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, - GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL); - } - gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse), - GST_BASE_PARSE_SRC_PAD (mp3parse), taglist); - g_free (codec); - - /* also signals the end of first-frame processing */ - mp3parse->sent_codec_tag = TRUE; - } - - /* we will create a taglist (if any of the parameters has changed) - * to add the tags that changed */ - taglist = NULL; - if (mp3parse->last_posted_crc != mp3parse->last_crc) { - gboolean using_crc; - - if (!taglist) { - taglist = gst_tag_list_new (); - } - mp3parse->last_posted_crc = mp3parse->last_crc; - if (mp3parse->last_posted_crc == CRC_PROTECTED) { - using_crc = TRUE; - } else { - using_crc = FALSE; - } - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC, - using_crc, NULL); - } - - if (mp3parse->last_posted_channel_mode != mp3parse->last_mode) { - if (!taglist) { - taglist = gst_tag_list_new (); - } - mp3parse->last_posted_channel_mode = mp3parse->last_mode; - - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE, - gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL); - } - - /* if the taglist exists, we need to send it */ - if (taglist) { - gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse), - GST_BASE_PARSE_SRC_PAD (mp3parse), taglist); - } - - /* usual clipping applies */ - frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP; - - return GST_FLOW_OK; -} diff --git a/gst/audioparsers/gstmpegaudioparse.h b/gst/audioparsers/gstmpegaudioparse.h deleted file mode 100644 index 68b2597..0000000 --- a/gst/audioparsers/gstmpegaudioparse.h +++ /dev/null @@ -1,111 +0,0 @@ -/* GStreamer MPEG audio parser - * Copyright (C) 2006-2007 Jan Schmidt - * Copyright (C) 2010 Mark Nauwelaerts - * Copyright (C) 2010 Nokia Corporation. All rights reserved. - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_MPEG_AUDIO_PARSE_H__ -#define __GST_MPEG_AUDIO_PARSE_H__ - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_MPEG_AUDIO_PARSE \ - (gst_mpeg_audio_parse_get_type()) -#define GST_MPEG_AUDIO_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_MPEG_AUDIO_PARSE, GstMpegAudioParse)) -#define GST_MPEG_AUDIO_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_MPEG_AUDIO_PARSE, GstMpegAudioParseClass)) -#define GST_IS_MPEG_AUDIO_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_MPEG_AUDIO_PARSE)) -#define GST_IS_MPEG_AUDIO_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_MPEG_AUDIO_PARSE)) - -typedef struct _GstMpegAudioParse GstMpegAudioParse; -typedef struct _GstMpegAudioParseClass GstMpegAudioParseClass; - -/** - * GstMpegAudioParse: - * - * The opaque GstMpegAudioParse object - */ -struct _GstMpegAudioParse { - GstBaseParse baseparse; - - /*< private >*/ - gint rate; - gint channels; - gint layer; - gint version; - - GstClockTime max_bitreservoir; - /* samples per frame */ - gint spf; - - gboolean sent_codec_tag; - guint last_posted_bitrate; - gint last_posted_crc, last_crc; - guint last_posted_channel_mode, last_mode; - - /* Bitrate from non-vbr headers */ - guint32 hdr_bitrate; - - /* Xing info */ - guint32 xing_flags; - guint32 xing_frames; - GstClockTime xing_total_time; - guint32 xing_bytes; - /* percent -> filepos mapping */ - guchar xing_seek_table[100]; - /* filepos -> percent mapping */ - guint16 xing_seek_table_inverse[256]; - guint32 xing_vbr_scale; - guint xing_bitrate; - - /* VBRI info */ - guint32 vbri_frames; - GstClockTime vbri_total_time; - guint32 vbri_bytes; - guint vbri_bitrate; - guint vbri_seek_points; - guint32 *vbri_seek_table; - gboolean vbri_valid; - - /* LAME info */ - guint32 encoder_delay; - guint32 encoder_padding; -}; - -/** - * GstMpegAudioParseClass: - * @parent_class: Element parent class. - * - * The opaque GstMpegAudioParseClass data structure. - */ -struct _GstMpegAudioParseClass { - GstBaseParseClass baseparse_class; -}; - -GType gst_mpeg_audio_parse_get_type (void); - -G_END_DECLS - -#endif /* __GST_MPEG_AUDIO_PARSE_H__ */ diff --git a/gst/audioparsers/plugin.c b/gst/audioparsers/plugin.c deleted file mode 100644 index 7d6d2f3..0000000 --- a/gst/audioparsers/plugin.c +++ /dev/null @@ -1,57 +0,0 @@ -/* GStreamer audio parsers - * Copyright (C) 2009 Tim-Philipp Müller - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstaacparse.h" -#include "gstamrparse.h" -#include "gstac3parse.h" -#include "gstdcaparse.h" -#include "gstflacparse.h" -#include "gstmpegaudioparse.h" - -static gboolean -plugin_init (GstPlugin * plugin) -{ - gboolean ret; - - ret = gst_element_register (plugin, "aacparse", - GST_RANK_PRIMARY + 1, GST_TYPE_AACPARSE); - ret &= gst_element_register (plugin, "amrparse", - GST_RANK_PRIMARY + 1, GST_TYPE_AMRPARSE); - ret &= gst_element_register (plugin, "ac3parse", - GST_RANK_PRIMARY + 1, GST_TYPE_AC3_PARSE); - ret &= gst_element_register (plugin, "dcaparse", - GST_RANK_PRIMARY + 1, GST_TYPE_DCA_PARSE); - ret &= gst_element_register (plugin, "flacparse", - GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE); - ret &= gst_element_register (plugin, "mpegaudioparse", - GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE); - - return ret; -} - - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "audioparsersbad", - "audioparsers", - plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 293e971..2f8207a 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -154,15 +154,11 @@ check_PROGRAMS = \ $(check_ofa) \ $(check_timidity) \ $(check_kate) \ - elements/aacparse \ - elements/ac3parse \ - elements/amrparse \ elements/autoconvert \ elements/autovideoconvert \ elements/asfmux \ elements/camerabin \ elements/dataurisrc \ - elements/flacparse \ elements/legacyresample \ $(check_jifmux) \ elements/jpegparse \ @@ -171,7 +167,6 @@ check_PROGRAMS = \ elements/mxfdemux \ elements/mxfmux \ elements/id3mux \ - elements/mpegaudioparse \ pipelines/mxf \ $(check_mimic) \ elements/rtpmux \ @@ -237,23 +232,6 @@ elements_rtpmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-0.10 $(GST_BASE_LIBS) elements_assrender_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS) elements_assrender_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-0.10 -lgstapp-0.10 $(GST_BASE_LIBS) $(LDADD) -# parser unit test convenience lib -noinst_LTLIBRARIES = libparser.la -libparser_la_SOURCES = elements/parser.c elements/parser.h -libparser_la_CFLAGS = \ - -I$(top_srcdir)/tests/check \ - $(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS) - -elements_aacparse_LDADD = libparser.la $(LDADD) - -elements_ac3parse_LDADD = libparser.la $(LDADD) - -elements_amrparse_LDADD = libparser.la $(LDADD) - -elements_flacparse_LDADD = libparser.la $(LDADD) - -elements_mpegaudioparse_LDADD = libparser.la $(LDADD) - EXTRA_DIST = gst-plugins-bad.supp orc_cog_CFLAGS = $(ORC_CFLAGS) diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index 8a74823..df8ab17 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -1,7 +1,4 @@ .dirstamp -aacparse -ac3parse -amrparse asfmux assrender autoconvert @@ -12,7 +9,6 @@ deinterleave dataurisrc faac faad -flacparse gdpdepay gdppay id3mux @@ -22,8 +18,8 @@ jifmux jpegparse kate legacyresample +logoinsert mpeg2enc -mpegaudioparse mplex mxfdemux mxfmux diff --git a/tests/check/elements/aacparse.c b/tests/check/elements/aacparse.c deleted file mode 100644 index af10a27..0000000 --- a/tests/check/elements/aacparse.c +++ /dev/null @@ -1,240 +0,0 @@ -/* - * GStreamer - * - * unit test for aacparse - * - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include -#include "parser.h" - -#define SRC_CAPS_CDATA "audio/mpeg, framed=(boolean)false, codec_data=(buffer)1190" -#define SRC_CAPS_TMPL "audio/mpeg, framed=(boolean)false, mpegversion=(int){2,4}" - -#define SINK_CAPS \ - "audio/mpeg, framed=(boolean)true" -#define SINK_CAPS_MPEG2 \ - "audio/mpeg, framed=(boolean)true, mpegversion=2, rate=48000, channels=2" -#define SINK_CAPS_MPEG4 \ - "audio/mpeg, framed=(boolean)true, mpegversion=4, rate=96000, channels=2" -#define SINK_CAPS_TMPL "audio/mpeg, framed=(boolean)true, mpegversion=(int){2,4}" - -GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_TMPL) - ); - -GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_TMPL) - ); - -/* some data */ -static guint8 adif_header[] = { - 'A', 'D', 'I', 'F' -}; - -static guint8 adts_frame_mpeg2[] = { - 0xff, 0xf9, 0x4c, 0x80, 0x01, 0xff, 0xfc, 0x21, 0x10, 0xd3, 0x20, 0x0c, - 0x32, 0x00, 0xc7 -}; - -static guint8 adts_frame_mpeg4[] = { - 0xff, 0xf1, 0x4c, 0x80, 0x01, 0xff, 0xfc, 0x21, 0x10, 0xd3, 0x20, 0x0c, - 0x32, 0x00, 0xc7 -}; - -static guint8 garbage_frame[] = { - 0xff, 0xff, 0xff, 0xff, 0xff -}; - -/* - * Test if the parser pushes data with ADIF header properly and detects the - * stream to MPEG4 properly. - */ -GST_START_TEST (test_parse_adif_normal) -{ - GstParserTest ptest; - - /* ADIF header */ - gst_parser_test_init (&ptest, adif_header, sizeof (adif_header), 1); - /* well, no garbage, followed by random data */ - ptest.series[2].size = 100; - ptest.series[2].num = 3; - /* and we do not really expect output frames */ - ptest.framed = FALSE; - /* Check that the negotiated caps are as expected */ - /* For ADIF parser assumes that data is always version 4 */ - ptest.sink_caps = - gst_caps_from_string (SINK_CAPS_MPEG4 ", stream-format=(string)adif"); - - gst_parser_test_run (&ptest, NULL); - - gst_caps_unref (ptest.sink_caps); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_adts_normal) -{ - gst_parser_test_normal (adts_frame_mpeg4, sizeof (adts_frame_mpeg4)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_adts_drain_single) -{ - gst_parser_test_drain_single (adts_frame_mpeg4, sizeof (adts_frame_mpeg4)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_adts_drain_garbage) -{ - gst_parser_test_drain_garbage (adts_frame_mpeg4, sizeof (adts_frame_mpeg4), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_adts_split) -{ - gst_parser_test_split (adts_frame_mpeg4, sizeof (adts_frame_mpeg4)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_adts_skip_garbage) -{ - gst_parser_test_skip_garbage (adts_frame_mpeg4, sizeof (adts_frame_mpeg4), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -/* - * Test if the src caps are set according to stream format (MPEG version). - */ -GST_START_TEST (test_parse_adts_detect_mpeg_version) -{ - gst_parser_test_output_caps (adts_frame_mpeg2, sizeof (adts_frame_mpeg2), - NULL, SINK_CAPS_MPEG2 ", stream-format=(string)adts"); -} - -GST_END_TEST; - -#define structure_get_int(s,f) \ - (g_value_get_int(gst_structure_get_value(s,f))) -#define fail_unless_structure_field_int_equals(s,field,num) \ - fail_unless_equals_int (structure_get_int(s,field), num) -/* - * Test if the parser handles raw stream and codec_data info properly. - */ -GST_START_TEST (test_parse_handle_codec_data) -{ - GstCaps *caps; - GstStructure *s; - const gchar *stream_format; - - /* Push random data. It should get through since the parser should be - * initialized because it got codec_data in the caps */ - caps = gst_parser_test_get_output_caps (NULL, 100, SRC_CAPS_CDATA); - fail_unless (caps != NULL); - - /* Check that the negotiated caps are as expected */ - /* When codec_data is present, parser assumes that data is version 4 */ - GST_LOG ("aac output caps: %" GST_PTR_FORMAT, caps); - s = gst_caps_get_structure (caps, 0); - fail_unless (gst_structure_has_name (s, "audio/mpeg")); - fail_unless_structure_field_int_equals (s, "mpegversion", 4); - fail_unless_structure_field_int_equals (s, "channels", 2); - fail_unless_structure_field_int_equals (s, "rate", 48000); - fail_unless (gst_structure_has_field (s, "codec_data")); - fail_unless (gst_structure_has_field (s, "stream-format")); - stream_format = gst_structure_get_string (s, "stream-format"); - fail_unless (strcmp (stream_format, "raw") == 0); - - gst_caps_unref (caps); -} - -GST_END_TEST; - - -static Suite * -aacparse_suite (void) -{ - Suite *s = suite_create ("aacparse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - /* ADIF tests */ - tcase_add_test (tc_chain, test_parse_adif_normal); - - /* ADTS tests */ - tcase_add_test (tc_chain, test_parse_adts_normal); - tcase_add_test (tc_chain, test_parse_adts_drain_single); - tcase_add_test (tc_chain, test_parse_adts_drain_garbage); - tcase_add_test (tc_chain, test_parse_adts_split); - tcase_add_test (tc_chain, test_parse_adts_skip_garbage); - tcase_add_test (tc_chain, test_parse_adts_detect_mpeg_version); - - /* Other tests */ - tcase_add_test (tc_chain, test_parse_handle_codec_data); - - return s; -} - - -/* - * TODO: - * - Both push- and pull-modes need to be tested - * * Pull-mode & EOS - */ - -int -main (int argc, char **argv) -{ - int nf; - - Suite *s = aacparse_suite (); - SRunner *sr = srunner_create (s); - - gst_check_init (&argc, &argv); - - /* init test context */ - ctx_factory = "aacparse"; - ctx_sink_template = &sinktemplate; - ctx_src_template = &srctemplate; - - srunner_run_all (sr, CK_NORMAL); - nf = srunner_ntests_failed (sr); - srunner_free (sr); - - return nf; -} diff --git a/tests/check/elements/ac3parse.c b/tests/check/elements/ac3parse.c deleted file mode 100644 index 03e8e1d..0000000 --- a/tests/check/elements/ac3parse.c +++ /dev/null @@ -1,163 +0,0 @@ -/* - * GStreamer - * - * unit test for ac3parse - * - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include -#include "parser.h" - -#define SRC_CAPS_TMPL "audio/x-ac3, framed=(boolean)false" -#define SINK_CAPS_TMPL "audio/x-ac3, framed=(boolean)true" - -GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_TMPL) - ); - -GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_TMPL) - ); - -/* some data */ - -static guint8 ac3_frame[512] = { - 0x0b, 0x77, 0xb6, 0xa8, 0x10, 0x40, 0x2f, 0x84, - 0x29, 0xcb, 0xfe, 0x75, 0x7c, 0xf9, 0xf3, 0xe7, - 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, - 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, - 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, - 0x7c, 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, - 0xf9, 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, - 0xf3, 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, - 0xe7, 0xcf, 0x9f, 0x3e, 0x7c, 0xf9, 0xf3, 0xe7, - 0xcf, 0x9f, 0x3e, 0x32, 0xd3, 0xff, 0xc0, 0x06, - 0xe9, 0x40, 0x00, 0x6e, 0x94, 0x00, 0x06, 0xe9, - 0x40, 0x00, 0x6e, 0x94, 0x00, 0x06, 0xe9, 0x40, - 0x00, 0x6e, 0x90, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, -}; - -static guint8 garbage_frame[] = { - 0xff, 0xff, 0xff, 0xff, 0xff -}; - - -GST_START_TEST (test_parse_normal) -{ - gst_parser_test_normal (ac3_frame, sizeof (ac3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_drain_single) -{ - gst_parser_test_drain_single (ac3_frame, sizeof (ac3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_drain_garbage) -{ - gst_parser_test_drain_garbage (ac3_frame, sizeof (ac3_frame), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_split) -{ - gst_parser_test_split (ac3_frame, sizeof (ac3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_skip_garbage) -{ - gst_parser_test_skip_garbage (ac3_frame, sizeof (ac3_frame), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_detect_stream) -{ - gst_parser_test_output_caps (ac3_frame, sizeof (ac3_frame), - NULL, SINK_CAPS_TMPL ",channels=1,rate=48000"); -} - -GST_END_TEST; - - -static Suite * -ac3parse_suite (void) -{ - Suite *s = suite_create ("ac3parse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - tcase_add_test (tc_chain, test_parse_normal); - tcase_add_test (tc_chain, test_parse_drain_single); - tcase_add_test (tc_chain, test_parse_drain_garbage); - tcase_add_test (tc_chain, test_parse_split); - tcase_add_test (tc_chain, test_parse_skip_garbage); - tcase_add_test (tc_chain, test_parse_detect_stream); - - return s; -} - - -/* - * TODO: - * - Both push- and pull-modes need to be tested - * * Pull-mode & EOS - */ - -int -main (int argc, char **argv) -{ - int nf; - - Suite *s = ac3parse_suite (); - SRunner *sr = srunner_create (s); - - gst_check_init (&argc, &argv); - - /* init test context */ - ctx_factory = "ac3parse"; - ctx_sink_template = &sinktemplate; - ctx_src_template = &srctemplate; - - srunner_run_all (sr, CK_NORMAL); - nf = srunner_ntests_failed (sr); - srunner_free (sr); - - return nf; -} diff --git a/tests/check/elements/amrparse.c b/tests/check/elements/amrparse.c deleted file mode 100644 index e5d64ca..0000000 --- a/tests/check/elements/amrparse.c +++ /dev/null @@ -1,327 +0,0 @@ -/* - * GStreamer - * - * unit test for amrparse - * - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include -#include "parser.h" - -#define SRC_CAPS_NB "audio/x-amr-nb-sh" -#define SRC_CAPS_WB "audio/x-amr-wb-sh" -#define SRC_CAPS_ANY "ANY" - -#define SINK_CAPS_NB "audio/AMR, rate=8000 , channels=1" -#define SINK_CAPS_WB "audio/AMR-WB, rate=16000 , channels=1" -#define SINK_CAPS_ANY "ANY" - -#define AMR_FRAME_DURATION (GST_SECOND/50) - -static GstStaticPadTemplate sinktemplate_nb = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_NB) - ); - -static GstStaticPadTemplate sinktemplate_wb = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_WB) - ); - -static GstStaticPadTemplate srctemplate_nb = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_NB) - ); - -static GstStaticPadTemplate srctemplate_wb = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_WB) - ); - - -/* some data */ - -static guint8 frame_data_nb[] = { - 0x0c, 0x56, 0x3c, 0x52, 0xe0, 0x61, 0xbc, 0x45, - 0x0f, 0x98, 0x2e, 0x01, 0x42, 0x02 -}; - -static guint8 frame_data_wb[] = { - 0x08, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, - 0x07, 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16 -}; - -static guint8 frame_hdr_nb[] = { - '#', '!', 'A', 'M', 'R', '\n' -}; - -static guint8 frame_hdr_wb[] = { - '#', '!', 'A', 'M', 'R', '-', 'W', 'B', '\n' -}; - -static guint8 garbage_frame[] = { - 0xff, 0xff, 0xff, 0xff, 0xff -}; - - -GST_START_TEST (test_parse_nb_normal) -{ - gst_parser_test_normal (frame_data_nb, sizeof (frame_data_nb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_nb_drain_single) -{ - gst_parser_test_drain_single (frame_data_nb, sizeof (frame_data_nb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_nb_drain_garbage) -{ - gst_parser_test_drain_garbage (frame_data_nb, sizeof (frame_data_nb), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_nb_split) -{ - gst_parser_test_split (frame_data_nb, sizeof (frame_data_nb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_nb_skip_garbage) -{ - gst_parser_test_skip_garbage (frame_data_nb, sizeof (frame_data_nb), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_nb_detect_stream) -{ - GstParserTest ptest; - GstCaps *old_ctx_caps; - - /* no input caps, override ctx */ - old_ctx_caps = ctx_input_caps; - ctx_input_caps = NULL; - - /* AMR-NB header */ - gst_parser_test_init (&ptest, frame_hdr_nb, sizeof (frame_hdr_nb), 1); - /* well, no garbage, followed by real data */ - ptest.series[2].data = frame_data_nb; - ptest.series[2].size = sizeof (frame_data_nb); - ptest.series[2].num = 10; - /* header gets dropped, so ... */ - /* buffer count will not match */ - ptest.framed = FALSE; - /* total size a bit less */ - ptest.dropped = sizeof (frame_hdr_nb); - - /* Check that the negotiated caps are as expected */ - ptest.sink_caps = gst_caps_from_string (SINK_CAPS_NB); - - gst_parser_test_run (&ptest, NULL); - - gst_caps_unref (ptest.sink_caps); - - ctx_input_caps = old_ctx_caps; -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_normal) -{ - gst_parser_test_normal (frame_data_wb, sizeof (frame_data_wb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_drain_single) -{ - gst_parser_test_drain_single (frame_data_wb, sizeof (frame_data_wb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_drain_garbage) -{ - gst_parser_test_drain_garbage (frame_data_wb, sizeof (frame_data_wb), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_split) -{ - gst_parser_test_split (frame_data_wb, sizeof (frame_data_wb)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_skip_garbage) -{ - gst_parser_test_skip_garbage (frame_data_wb, sizeof (frame_data_wb), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_wb_detect_stream) -{ - GstParserTest ptest; - GstCaps *old_ctx_caps; - - /* no input caps, override ctx */ - old_ctx_caps = ctx_input_caps; - ctx_input_caps = NULL; - - /* AMR-WB header */ - gst_parser_test_init (&ptest, frame_hdr_wb, sizeof (frame_hdr_wb), 1); - /* well, no garbage, followed by real data */ - ptest.series[2].data = frame_data_wb; - ptest.series[2].size = sizeof (frame_data_wb); - ptest.series[2].num = 10; - /* header gets dropped, so ... */ - /* buffer count will not match */ - ptest.framed = FALSE; - /* total size a bit less */ - ptest.dropped = sizeof (frame_hdr_wb); - - /* Check that the negotiated caps are as expected */ - ptest.sink_caps = gst_caps_from_string (SINK_CAPS_WB); - - gst_parser_test_run (&ptest, NULL); - - gst_caps_unref (ptest.sink_caps); - - ctx_input_caps = old_ctx_caps; -} - -GST_END_TEST; - - - -/* - * Create test suite. - */ -static Suite * -amrnb_parse_suite (void) -{ - Suite *s = suite_create ("amrwb_parse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - /* AMR-NB tests */ - tcase_add_test (tc_chain, test_parse_nb_normal); - tcase_add_test (tc_chain, test_parse_nb_drain_single); - tcase_add_test (tc_chain, test_parse_nb_drain_garbage); - tcase_add_test (tc_chain, test_parse_nb_split); - tcase_add_test (tc_chain, test_parse_nb_detect_stream); - tcase_add_test (tc_chain, test_parse_nb_skip_garbage); - - return s; -} - -static Suite * -amrwb_parse_suite (void) -{ - Suite *s = suite_create ("amrnb_parse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - /* AMR-WB tests */ - tcase_add_test (tc_chain, test_parse_wb_normal); - tcase_add_test (tc_chain, test_parse_wb_drain_single); - tcase_add_test (tc_chain, test_parse_wb_drain_garbage); - tcase_add_test (tc_chain, test_parse_wb_split); - tcase_add_test (tc_chain, test_parse_wb_detect_stream); - tcase_add_test (tc_chain, test_parse_wb_skip_garbage); - - return s; -} - -/* - * TODO: - * - Both push- and pull-modes need to be tested - * * Pull-mode & EOS - */ - -int -main (int argc, char **argv) -{ - int nf; - GstCaps *caps; - - Suite *s = amrnb_parse_suite (); - SRunner *sr = srunner_create (s); - - gst_check_init (&argc, &argv); - - /* init test context */ - ctx_factory = "amrparse"; - ctx_sink_template = &sinktemplate_nb; - ctx_src_template = &srctemplate_nb; - caps = gst_caps_from_string (SRC_CAPS_NB); - g_assert (caps); - ctx_input_caps = caps; - - srunner_run_all (sr, CK_NORMAL); - nf = srunner_ntests_failed (sr); - srunner_free (sr); - gst_caps_unref (caps); - - s = amrwb_parse_suite (); - sr = srunner_create (s); - - ctx_sink_template = &sinktemplate_wb; - ctx_src_template = &srctemplate_wb; - caps = gst_caps_from_string (SRC_CAPS_WB); - g_assert (caps); - ctx_input_caps = caps; - - srunner_run_all (sr, CK_NORMAL); - nf += srunner_ntests_failed (sr); - srunner_free (sr); - gst_caps_unref (caps); - - return nf; -} diff --git a/tests/check/elements/flacparse.c b/tests/check/elements/flacparse.c deleted file mode 100644 index 0c25bc6..0000000 --- a/tests/check/elements/flacparse.c +++ /dev/null @@ -1,299 +0,0 @@ -/* - * GStreamer - * - * unit test for flacparse - * - * Copyright (C) 2010 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include -#include "parser.h" - -#define SRC_CAPS_TMPL "audio/x-flac, framed=(boolean)false" -#define SINK_CAPS_TMPL "audio/x-flac, framed=(boolean)true" - -GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_TMPL) - ); - -GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_TMPL) - ); - -/* some data */ -static guint8 streaminfo_header[] = { - 0x7f, 0x46, 0x4c, 0x41, 0x43, 0x01, 0x00, 0x00, - 0x02, 0x66, 0x4c, 0x61, 0x43, 0x00, 0x00, 0x00, - 0x22, 0x12, 0x00, 0x12, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x0a, 0xc4, 0x40, 0xf0, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00 -}; - -static guint8 comment_header[] = { - 0x84, 0x00, 0x00, 0x08, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00 -}; - -static guint8 flac_frame[] = { - 0xff, 0xf8, 0xa9, 0x08, 0x00, 0x50, 0x18, 0x06, - 0x6a, 0x0c, 0xce, 0x13, 0x24, 0x19, 0x68, 0x00, - 0x46, 0x23, 0x08, 0xca, 0xcb, 0x58, 0x9c, 0x26, - 0x92, 0x30, 0xa6, 0x29, 0x8a, 0xca, 0xd1, 0x18, - 0xae, 0x26, 0x5c, 0x90, 0x60, 0xbf, 0x11, 0xad, - 0x43, 0x02, 0x06, 0x26, 0xbd, 0x35, 0xdd, 0xa3, - 0x11, 0xa6, 0x4d, 0x18, 0x8c, 0x9a, 0xe4, 0x62, - 0xd9, 0x23, 0x11, 0x8b, 0xcb, 0x56, 0x55, 0x45, - 0xc2, 0x18, 0x56, 0xa2, 0xe2, 0xe1, 0x18, 0x99, - 0x54, 0x98, 0x46, 0x4d, 0x08, 0x70, 0x9a, 0x64, - 0xc4, 0x18, 0x4f, 0x27, 0x64, 0x31, 0x66, 0x27, - 0x79, 0x19, 0x3c, 0x8c, 0x8c, 0xa3, 0x44, 0x18, - 0x23, 0xd2, 0x6b, 0x8b, 0x64, 0x8c, 0x21, 0x84, - 0xd6, 0x23, 0x13, 0x13, 0x2d, 0x44, 0xca, 0x5a, - 0x23, 0x09, 0x93, 0x25, 0x18, 0x10, 0x61, 0x38, - 0xb4, 0x60, 0x8f, 0x2c, 0x8d, 0x26, 0xb4, 0xc9, - 0xd9, 0x19, 0x19, 0x34, 0xd7, 0x31, 0x06, 0x10, - 0xc4, 0x30, 0x83, 0x17, 0xe2, 0x0c, 0x2c, 0xc4, - 0xc8, 0xc9, 0x3c, 0x5e, 0x93, 0x11, 0x8a, 0x62, - 0x64, 0x8c, 0x26, 0x23, 0x22, 0x30, 0x9a, 0x58, - 0x86, 0x04, 0x18, 0x4c, 0xab, 0x2b, 0x26, 0x5c, - 0x46, 0x88, 0xcb, 0xb1, 0x0d, 0x26, 0xbb, 0x5e, - 0x8c, 0xa7, 0x64, 0x31, 0x3d, 0x31, 0x06, 0x26, - 0x43, 0x17, 0xa3, 0x08, 0x61, 0x06, 0x17, 0xc4, - 0x62, 0xec, 0x4d, 0x4b, 0x2e, 0x2d, 0x4a, 0x94, - 0xa4, 0xc2, 0x31, 0x4c, 0x4c, 0x20, 0xc0, 0x83, - 0x14, 0x8c, 0x27, 0x8b, 0x31, 0x23, 0x2f, 0x23, - 0x11, 0x91, 0x94, 0x65, 0x1a, 0x20, 0xc2, 0x18, - 0x86, 0x51, 0x88, 0x62, 0x7c, 0x43, 0x2e, 0xa3, - 0x04, 0x18, 0x8c, 0x20, 0xc2, 0xf5, 0xaa, 0x94, - 0xc2, 0x31, 0x32, 0xd2, 0xb2, 0xa2, 0x30, 0xba, - 0x10, 0xc2, 0xb5, 0x89, 0xa5, 0x18, 0x10, 0x62, - 0x9a, 0x10, 0x61, 0x19, 0x72, 0x71, 0x1a, 0xb9, - 0x0c, 0x23, 0x46, 0x10, 0x62, 0x78, 0x81, 0x82, - 0x3d, 0x75, 0xea, 0x6b, 0x51, 0x8b, 0x61, 0x06, - 0x08, 0x62, 0x32, 0x5e, 0x84, 0x18, 0x27, 0x25, - 0xc2, 0x6a, 0x4b, 0x51, 0x31, 0x34, 0x5e, 0x29, - 0xa1, 0x3c, 0x4d, 0x26, 0x23, 0x10, 0xc2, 0x6b, - 0xb1, 0x0d, 0x11, 0xae, 0x46, 0x88, 0x31, 0x35, - 0xb1, 0x06, 0x08, 0x79, 0x7e, 0x4f, 0x53, 0x23, - 0x29, 0xa4, 0x30, 0x20, 0x30, 0x23, 0x5a, 0xb2, - 0xc8, 0x60, 0x9c, 0x93, 0x13, 0x17, 0x92, 0x98, - 0x46, 0x13, 0x54, 0x53, 0x08, 0xcb, 0x13, 0xa1, - 0x1a, 0x89, 0xe5, 0x46, 0x08, 0x18, 0x10, 0x30, - 0x9d, 0x68, 0xc2, 0x1c, 0x46, 0x46, 0xae, 0x62, - 0x1a, 0x46, 0x4e, 0x4d, 0x34, 0x8c, 0xbd, 0x26, - 0xc0, 0x40, 0x62, 0xc9, 0xa9, 0x31, 0x74, 0xa8, - 0x99, 0x52, 0xb0, 0x8c, 0xa9, 0x29, 0x84, 0x61, - 0x19, 0x54, 0x43, 0x02, 0x06, 0x04, 0x32, 0xe5, - 0x18, 0x21, 0x91, 0x8b, 0xf2, 0xcc, 0x10, 0x30, - 0x8e, 0x23, 0xc4, 0x76, 0x43, 0x08, 0x30, 0x83, - 0x08, 0x62, 0x6c, 0x4e, 0xe2, 0x35, 0x96, 0xd0, - 0x8e, 0x89, 0x97, 0x42, 0x18, 0x91, 0x84, 0x61, - 0x3c, 0x26, 0xa5, 0x2c, 0x4e, 0x17, 0x94, 0xb8, - 0xb5, 0xa4, 0xcb, 0x88, 0xc9, 0x84, 0x18, 0xb9, - 0x84, 0x19, 0x23, 0x2d, 0xa4, 0x64, 0x62, 0x18, - 0x86, 0x53, 0x93, 0xcb, 0x30, 0x8f, 0x2f, 0x93, - 0x55, 0xc4, 0xd7, 0x08, 0x62, 0xb8, 0x46, 0x84, - 0x68, 0xa3, 0x02, 0xaf, 0x33 -}; - -static guint8 garbage_frame[] = { - 0xff, 0xff, 0xff, 0xff, 0xff -}; - - -GST_START_TEST (test_parse_flac_normal) -{ - gst_parser_test_normal (flac_frame, sizeof (flac_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_flac_drain_single) -{ - gst_parser_test_drain_single (flac_frame, sizeof (flac_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_flac_drain_garbage) -{ - /* We always output the after frame garbage too because we - * have no way of detecting it - */ -#if 0 - gst_parser_test_drain_garbage (flac_frame, sizeof (flac_frame), - garbage_frame, sizeof (garbage_frame)); -#endif - guint8 frame[sizeof (flac_frame) + sizeof (garbage_frame)]; - - memcpy (frame, flac_frame, sizeof (flac_frame)); - memcpy (frame + sizeof (flac_frame), garbage_frame, sizeof (garbage_frame)); - - gst_parser_test_drain_single (frame, sizeof (frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_flac_split) -{ - gst_parser_test_split (flac_frame, sizeof (flac_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_flac_skip_garbage) -{ - /* We always include the garbage into the frame because - * we have no easy way for finding the real end of the - * frame. The decoder will later skip the garbage - */ -#if 0 - gst_parser_test_skip_garbage (flac_frame, sizeof (flac_frame), - garbage_frame, sizeof (garbage_frame)); -#endif - guint8 frame[sizeof (flac_frame) + sizeof (garbage_frame)]; - - memcpy (frame, flac_frame, sizeof (flac_frame)); - memcpy (frame + sizeof (flac_frame), garbage_frame, sizeof (garbage_frame)); - - gst_parser_test_normal (frame, sizeof (frame)); -} - -GST_END_TEST; - - -#define structure_get_int(s,f) \ - (g_value_get_int(gst_structure_get_value(s,f))) -#define fail_unless_structure_field_int_equals(s,field,num) \ - fail_unless_equals_int (structure_get_int(s,field), num) -/* - * Test if the parser handles raw stream and codec_data info properly. - */ -GST_START_TEST (test_parse_flac_detect_stream) -{ - GstCaps *caps; - GstStructure *s; - const GValue *streamheader; - GArray *bufarr; - gint i; - - /* Push random data. It should get through since the parser should be - * initialized because it got codec_data in the caps */ - caps = gst_parser_test_get_output_caps (flac_frame, sizeof (flac_frame), - SRC_CAPS_TMPL); - fail_unless (caps != NULL); - - /* Check that the negotiated caps are as expected */ - /* When codec_data is present, parser assumes that data is version 4 */ - GST_LOG ("flac output caps: %" GST_PTR_FORMAT, caps); - s = gst_caps_get_structure (caps, 0); - fail_unless (gst_structure_has_name (s, "audio/x-flac")); - fail_unless_structure_field_int_equals (s, "channels", 1); - fail_unless_structure_field_int_equals (s, "rate", 44100); - fail_unless (gst_structure_has_field (s, "streamheader")); - streamheader = gst_structure_get_value (s, "streamheader"); - fail_unless (G_VALUE_TYPE (streamheader) == GST_TYPE_ARRAY); - bufarr = g_value_peek_pointer (streamheader); - fail_unless (bufarr->len == 2); - for (i = 0; i < bufarr->len; i++) { - GstBuffer *buf; - GValue *bufval = &g_array_index (bufarr, GValue, i); - - fail_unless (G_VALUE_TYPE (bufval) == GST_TYPE_BUFFER); - buf = g_value_peek_pointer (bufval); - if (i == 0) { - fail_unless (GST_BUFFER_SIZE (buf) == sizeof (streaminfo_header)); - fail_unless (memcmp (buf, streaminfo_header, sizeof (streaminfo_header))); - } else if (i == 1) { - fail_unless (GST_BUFFER_SIZE (buf) == sizeof (comment_header)); - fail_unless (memcmp (buf, comment_header, sizeof (comment_header))); - } - } - - gst_caps_unref (caps); -} - -GST_END_TEST; - - -static Suite * -flacparse_suite (void) -{ - Suite *s = suite_create ("flacparse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - tcase_add_test (tc_chain, test_parse_flac_normal); - tcase_add_test (tc_chain, test_parse_flac_drain_single); - tcase_add_test (tc_chain, test_parse_flac_drain_garbage); - tcase_add_test (tc_chain, test_parse_flac_split); - tcase_add_test (tc_chain, test_parse_flac_skip_garbage); - - /* Other tests */ - tcase_add_test (tc_chain, test_parse_flac_detect_stream); - - return s; -} - - -/* - * TODO: - * - Both push- and pull-modes need to be tested - * * Pull-mode & EOS - */ - -int -main (int argc, char **argv) -{ - int nf; - - Suite *s = flacparse_suite (); - SRunner *sr = srunner_create (s); - - gst_check_init (&argc, &argv); - - /* init test context */ - ctx_factory = "flacparse"; - ctx_sink_template = &sinktemplate; - ctx_src_template = &srctemplate; - ctx_discard = 3; - ctx_headers[0].data = streaminfo_header; - ctx_headers[0].size = sizeof (streaminfo_header); - ctx_headers[1].data = comment_header; - ctx_headers[1].size = sizeof (comment_header); - /* custom offsets, and ts always repeatedly 0 */ - ctx_no_metadata = TRUE; - - srunner_run_all (sr, CK_NORMAL); - nf = srunner_ntests_failed (sr); - srunner_free (sr); - - return nf; -} diff --git a/tests/check/elements/mpegaudioparse.c b/tests/check/elements/mpegaudioparse.c deleted file mode 100644 index 69a0864..0000000 --- a/tests/check/elements/mpegaudioparse.c +++ /dev/null @@ -1,172 +0,0 @@ -/* - * GStreamer - * - * unit test for aacparse - * - * Copyright (C) 2008 Nokia Corporation. All rights reserved. - * - * Contact: Stefan Kost - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include -#include "parser.h" - -#define SRC_CAPS_TMPL "audio/mpeg, parsed=(boolean)false, mpegversion=(int)1" -#define SINK_CAPS_TMPL "audio/mpeg, parsed=(boolean)true, mpegversion=(int)1" - -GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SINK_CAPS_TMPL) - ); - -GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS (SRC_CAPS_TMPL) - ); - -const gchar *factory = "aacparse"; - -/* some data */ -static guint8 mp3_frame[384] = { - 0xff, 0xfb, 0x94, 0xc4, 0xff, 0x83, 0xc0, 0x00, - 0x01, 0xa4, 0x00, 0x00, 0x00, 0x20, 0x00, 0x00, - 0x34, 0x80, 0x00, 0x00, 0x04, 0x00, -}; - -static guint8 garbage_frame[] = { - 0xff, 0xff, 0xff, 0xff, 0xff -}; - - -GST_START_TEST (test_parse_normal) -{ - gst_parser_test_normal (mp3_frame, sizeof (mp3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_drain_single) -{ - gst_parser_test_drain_single (mp3_frame, sizeof (mp3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_drain_garbage) -{ - gst_parser_test_drain_garbage (mp3_frame, sizeof (mp3_frame), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_split) -{ - gst_parser_test_split (mp3_frame, sizeof (mp3_frame)); -} - -GST_END_TEST; - - -GST_START_TEST (test_parse_skip_garbage) -{ - gst_parser_test_skip_garbage (mp3_frame, sizeof (mp3_frame), - garbage_frame, sizeof (garbage_frame)); -} - -GST_END_TEST; - - -#define structure_get_int(s,f) \ - (g_value_get_int(gst_structure_get_value(s,f))) -#define fail_unless_structure_field_int_equals(s,field,num) \ - fail_unless_equals_int (structure_get_int(s,field), num) - -GST_START_TEST (test_parse_detect_stream) -{ - GstStructure *s; - GstCaps *caps; - - caps = gst_parser_test_get_output_caps (mp3_frame, sizeof (mp3_frame), NULL); - - fail_unless (caps != NULL); - - GST_LOG ("mpegaudio output caps: %" GST_PTR_FORMAT, caps); - s = gst_caps_get_structure (caps, 0); - fail_unless (gst_structure_has_name (s, "audio/mpeg")); - fail_unless_structure_field_int_equals (s, "mpegversion", 1); - fail_unless_structure_field_int_equals (s, "layer", 3); - fail_unless_structure_field_int_equals (s, "channels", 1); - fail_unless_structure_field_int_equals (s, "rate", 48000); - - gst_caps_unref (caps); -} - -GST_END_TEST; - - -static Suite * -mpegaudioparse_suite (void) -{ - Suite *s = suite_create ("mpegaudioparse"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - tcase_add_test (tc_chain, test_parse_normal); - tcase_add_test (tc_chain, test_parse_drain_single); - tcase_add_test (tc_chain, test_parse_drain_garbage); - tcase_add_test (tc_chain, test_parse_split); - tcase_add_test (tc_chain, test_parse_skip_garbage); - tcase_add_test (tc_chain, test_parse_detect_stream); - - return s; -} - - -/* - * TODO: - * - Both push- and pull-modes need to be tested - * * Pull-mode & EOS - */ - -int -main (int argc, char **argv) -{ - int nf; - - Suite *s = mpegaudioparse_suite (); - SRunner *sr = srunner_create (s); - - gst_check_init (&argc, &argv); - - /* init test context */ - ctx_factory = "mpegaudioparse"; - ctx_sink_template = &sinktemplate; - ctx_src_template = &srctemplate; - - srunner_run_all (sr, CK_NORMAL); - nf = srunner_ntests_failed (sr); - srunner_free (sr); - - return nf; -} -- 2.7.4