mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
The pre_push_frame default clipping behaviour was introduced in 2010 with commit30be03004e
and modified with commit4163969a24
in 2011, when most parsers didn't implement a pre_push_frame yet. Not having it meant that clipping was done by default. Those that did implement a pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag adjusted as part of the 2011 refactor work. All other parsers got a pre_push_frame vfunc implementation only in 2013, but seem to have forgot to keep the clipping behaviour, as was done automatically when a pre_push_frame implementation doesn't exist for the parser. aacparse lost it with commit91d4abcea
in July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting in commits6f89b430e
,d2ab5199b
,29f2cae12
,753d3c23a
and292780574
.
454 lines
13 KiB
C
454 lines
13 KiB
C
/* GStreamer Adaptive Multi-Rate parser plugin
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* Copyright (C) 2006 Edgard Lima <edgard.lima@gmail.com>
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* Copyright (C) 2008 Nokia Corporation. All rights reserved.
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*
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-amrparse
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* @title: amrparse
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* @short_description: AMR parser
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* @see_also: #GstAmrnbDec, #GstAmrnbEnc
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*
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* This is an AMR parser capable of handling both narrow-band and wideband
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* formats.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
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* ]|
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstamrparse.h"
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#include <gst/pbutils/pbutils.h>
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1;"
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"audio/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1;")
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);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-amr-nb-sh; audio/x-amr-wb-sh"));
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GST_DEBUG_CATEGORY_STATIC (amrparse_debug);
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#define GST_CAT_DEFAULT amrparse_debug
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static const gint block_size_nb[16] =
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{ 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
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static const gint block_size_wb[16] =
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{ 17, 23, 32, 36, 40, 46, 50, 58, 60, 5, -1, -1, -1, -1, 0, 0 };
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/* AMR has a "hardcoded" framerate of 50fps */
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#define AMR_FRAMES_PER_SECOND 50
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#define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND)
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#define AMR_MIME_HEADER_SIZE 9
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static gboolean gst_amr_parse_start (GstBaseParse * parse);
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static gboolean gst_amr_parse_stop (GstBaseParse * parse);
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static gboolean gst_amr_parse_sink_setcaps (GstBaseParse * parse,
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GstCaps * caps);
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static GstCaps *gst_amr_parse_sink_getcaps (GstBaseParse * parse,
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GstCaps * filter);
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static GstFlowReturn gst_amr_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize);
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static GstFlowReturn gst_amr_parse_pre_push_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
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/**
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* gst_amr_parse_class_init:
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* @klass: GstAmrParseClass.
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*
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*/
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static void
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gst_amr_parse_class_init (GstAmrParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0,
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"AMR-NB audio stream parser");
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (element_class,
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"AMR audio stream parser", "Codec/Parser/Audio",
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"Adaptive Multi-Rate audio parser",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
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parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_setcaps);
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parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_getcaps);
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parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amr_parse_handle_frame);
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parse_class->pre_push_frame =
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GST_DEBUG_FUNCPTR (gst_amr_parse_pre_push_frame);
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}
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/**
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* gst_amr_parse_init:
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* @amrparse: #GstAmrParse
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* @klass: #GstAmrParseClass.
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*
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*/
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static void
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gst_amr_parse_init (GstAmrParse * amrparse)
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{
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/* init rest */
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
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GST_DEBUG ("initialized");
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (amrparse));
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (amrparse));
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}
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/**
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* gst_amr_parse_set_src_caps:
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* @amrparse: #GstAmrParse.
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*
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* Set source pad caps according to current knowledge about the
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* audio stream.
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*
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* Returns: TRUE if caps were successfully set.
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*/
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static gboolean
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gst_amr_parse_set_src_caps (GstAmrParse * amrparse)
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{
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GstCaps *src_caps = NULL;
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gboolean res = FALSE;
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if (amrparse->wide) {
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GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-WB");
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src_caps = gst_caps_new_simple ("audio/AMR-WB",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 16000, NULL);
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} else {
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GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-NB");
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/* Max. size of NB frame is 31 bytes, so we can set the min. frame
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size to 32 (+1 for next frame header) */
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 32);
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src_caps = gst_caps_new_simple ("audio/AMR",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
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}
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gst_pad_use_fixed_caps (GST_BASE_PARSE (amrparse)->srcpad);
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res = gst_pad_set_caps (GST_BASE_PARSE (amrparse)->srcpad, src_caps);
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gst_caps_unref (src_caps);
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return res;
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}
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/**
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* gst_amr_parse_sink_setcaps:
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* @sinkpad: GstPad
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* @caps: GstCaps
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_amr_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
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{
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GstAmrParse *amrparse;
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GstStructure *structure;
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const gchar *name;
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amrparse = GST_AMR_PARSE (parse);
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structure = gst_caps_get_structure (caps, 0);
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name = gst_structure_get_name (structure);
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GST_DEBUG_OBJECT (amrparse, "setcaps: %s", name);
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if (!strncmp (name, "audio/x-amr-wb-sh", 17)) {
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amrparse->block_size = block_size_wb;
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amrparse->wide = 1;
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} else if (!strncmp (name, "audio/x-amr-nb-sh", 17)) {
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amrparse->block_size = block_size_nb;
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amrparse->wide = 0;
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} else {
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GST_WARNING ("Unknown caps");
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return FALSE;
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}
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amrparse->need_header = FALSE;
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gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);
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gst_amr_parse_set_src_caps (amrparse);
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return TRUE;
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}
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/**
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* gst_amr_parse_parse_header:
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* @amrparse: #GstAmrParse
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* @data: Header data to be parsed.
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* @skipsize: Output argument where the frame size will be stored.
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*
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* Check if the given data contains an AMR mime header.
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_amr_parse_parse_header (GstAmrParse * amrparse,
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const guint8 * data, gint * skipsize)
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{
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GST_DEBUG_OBJECT (amrparse, "Parsing header data");
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if (!memcmp (data, "#!AMR-WB\n", 9)) {
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GST_DEBUG_OBJECT (amrparse, "AMR-WB detected");
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amrparse->block_size = block_size_wb;
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amrparse->wide = TRUE;
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*skipsize = amrparse->header = 9;
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} else if (!memcmp (data, "#!AMR\n", 6)) {
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GST_DEBUG_OBJECT (amrparse, "AMR-NB detected");
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amrparse->block_size = block_size_nb;
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amrparse->wide = FALSE;
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*skipsize = amrparse->header = 6;
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} else
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return FALSE;
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gst_amr_parse_set_src_caps (amrparse);
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return TRUE;
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}
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/**
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* gst_amr_parse_check_valid_frame:
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* @parse: #GstBaseParse.
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* @buffer: #GstBuffer.
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* @framesize: Output variable where the found frame size is put.
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* @skipsize: Output variable which tells how much data needs to be skipped
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* until a frame header is found.
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*
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* Implementation of "check_valid_frame" vmethod in #GstBaseParse class.
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*
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* Returns: TRUE if the given data contains valid frame.
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*/
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static GstFlowReturn
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gst_amr_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize)
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{
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GstBuffer *buffer;
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GstMapInfo map;
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gint fsize = 0, mode, dsize;
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GstAmrParse *amrparse;
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GstFlowReturn ret = GST_FLOW_OK;
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gboolean found = FALSE;
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amrparse = GST_AMR_PARSE (parse);
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buffer = frame->buffer;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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dsize = map.size;
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GST_LOG ("buffer: %d bytes", dsize);
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if (amrparse->need_header) {
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if (dsize >= AMR_MIME_HEADER_SIZE &&
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gst_amr_parse_parse_header (amrparse, map.data, skipsize)) {
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amrparse->need_header = FALSE;
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gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);
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} else {
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GST_WARNING ("media doesn't look like a AMR format");
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}
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/* We return FALSE, so this frame won't get pushed forward. Instead,
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the "skip" value is set, so next time we will receive a valid frame. */
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goto done;
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}
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*skipsize = 1;
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/* Does this look like a possible frame header candidate? */
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if ((map.data[0] & 0x83) == 0) {
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/* Yep. Retrieve the frame size */
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mode = (map.data[0] >> 3) & 0x0F;
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fsize = amrparse->block_size[mode] + 1; /* +1 for the header byte */
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/* We recognize this data as a valid frame when:
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* - We are in sync. There is no need for extra checks then
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* - We are in EOS. There might not be enough data to check next frame
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* - Sync is lost, but the following data after this frame seem
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* to contain a valid header as well (and there is enough data to
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* perform this check)
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*/
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if (fsize) {
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*skipsize = 0;
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/* in sync, no further check */
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if (!GST_BASE_PARSE_LOST_SYNC (parse)) {
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found = TRUE;
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} else if (dsize > fsize) {
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/* enough data, check for next sync */
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if ((map.data[fsize] & 0x83) == 0)
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found = TRUE;
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} else if (GST_BASE_PARSE_DRAINING (parse)) {
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/* not enough, but draining, so ok */
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found = TRUE;
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}
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}
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}
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done:
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gst_buffer_unmap (buffer, &map);
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if (found && fsize <= map.size) {
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ret = gst_base_parse_finish_frame (parse, frame, fsize);
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}
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return ret;
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}
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/**
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* gst_amr_parse_start:
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* @parse: #GstBaseParse.
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*
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* Implementation of "start" vmethod in #GstBaseParse class.
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_amr_parse_start (GstBaseParse * parse)
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{
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GstAmrParse *amrparse;
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amrparse = GST_AMR_PARSE (parse);
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GST_DEBUG ("start");
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amrparse->need_header = TRUE;
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amrparse->header = 0;
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amrparse->sent_codec_tag = FALSE;
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return TRUE;
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}
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/**
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* gst_amr_parse_stop:
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* @parse: #GstBaseParse.
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*
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* Implementation of "stop" vmethod in #GstBaseParse class.
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_amr_parse_stop (GstBaseParse * parse)
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{
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GstAmrParse *amrparse;
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amrparse = GST_AMR_PARSE (parse);
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GST_DEBUG ("stop");
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amrparse->need_header = TRUE;
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amrparse->header = 0;
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return TRUE;
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}
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static GstCaps *
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gst_amr_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
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{
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GstCaps *peercaps, *templ;
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GstCaps *res;
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templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
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peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), filter);
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if (peercaps) {
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guint i, n;
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/* Rename structure names */
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peercaps = gst_caps_make_writable (peercaps);
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n = gst_caps_get_size (peercaps);
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for (i = 0; i < n; i++) {
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GstStructure *s = gst_caps_get_structure (peercaps, i);
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if (gst_structure_has_name (s, "audio/AMR"))
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gst_structure_set_name (s, "audio/x-amr-nb-sh");
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else
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gst_structure_set_name (s, "audio/x-amr-wb-sh");
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}
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res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (peercaps);
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res = gst_caps_make_writable (res);
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/* Append the template caps because we still want to accept
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* caps without any fields in the case upstream does not
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* know anything.
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*/
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gst_caps_append (res, templ);
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} else {
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res = templ;
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}
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (res);
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res = intersection;
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}
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return res;
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}
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static GstFlowReturn
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gst_amr_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
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{
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GstAmrParse *amrparse = GST_AMR_PARSE (parse);
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if (!amrparse->sent_codec_tag) {
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GstTagList *taglist;
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GstCaps *caps;
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/* codec tag */
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caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
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if (G_UNLIKELY (caps == NULL)) {
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if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
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GST_INFO_OBJECT (parse, "Src pad is flushing");
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return GST_FLOW_FLUSHING;
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} else {
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GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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taglist = gst_tag_list_new_empty ();
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gst_pb_utils_add_codec_description_to_tag_list (taglist,
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GST_TAG_AUDIO_CODEC, caps);
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gst_caps_unref (caps);
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gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
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gst_tag_list_unref (taglist);
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/* also signals the end of first-frame processing */
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amrparse->sent_codec_tag = TRUE;
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}
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frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
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return GST_FLOW_OK;
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}
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