mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
412 lines
12 KiB
C
412 lines
12 KiB
C
/* GStreamer Adaptive Multi-Rate parser plugin
|
|
* Copyright (C) 2006 Edgard Lima <edgard.lima@indt.org.br>
|
|
* Copyright (C) 2008 Nokia Corporation. All rights reserved.
|
|
*
|
|
* Contact: Stefan Kost <stefan.kost@nokia.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* <refsect2>
|
|
* <title>Example launch line</title>
|
|
* |[
|
|
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
|
|
* ]|
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
|
|
#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 (amrparse_debug);
|
|
#define GST_CAT_DEFAULT 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
|
|
|
|
static gboolean gst_amr_parse_start (GstBaseParse * parse);
|
|
static gboolean gst_amr_parse_stop (GstBaseParse * parse);
|
|
|
|
static gboolean gst_amr_parse_sink_setcaps (GstBaseParse * parse,
|
|
GstCaps * caps);
|
|
static GstCaps *gst_amr_parse_sink_getcaps (GstBaseParse * parse,
|
|
GstCaps * filter);
|
|
|
|
static GstFlowReturn gst_amr_parse_handle_frame (GstBaseParse * parse,
|
|
GstBaseParseFrame * frame, gint * skipsize);
|
|
|
|
G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
|
|
|
|
/**
|
|
* gst_amr_parse_class_init:
|
|
* @klass: GstAmrParseClass.
|
|
*
|
|
*/
|
|
static void
|
|
gst_amr_parse_class_init (GstAmrParseClass * klass)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0,
|
|
"AMR-NB audio stream parser");
|
|
|
|
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_static_metadata (element_class,
|
|
"AMR audio stream parser", "Codec/Parser/Audio",
|
|
"Adaptive Multi-Rate audio parser",
|
|
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
|
|
|
|
parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
|
|
parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
|
|
parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_setcaps);
|
|
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_getcaps);
|
|
parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amr_parse_handle_frame);
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_amr_parse_init:
|
|
* @amrparse: #GstAmrParse
|
|
* @klass: #GstAmrParseClass.
|
|
*
|
|
*/
|
|
static void
|
|
gst_amr_parse_init (GstAmrParse * amrparse)
|
|
{
|
|
/* init rest */
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
|
|
GST_DEBUG ("initialized");
|
|
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_amr_parse_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_amr_parse_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_amr_parse_sink_setcaps:
|
|
* @sinkpad: GstPad
|
|
* @caps: GstCaps
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
static gboolean
|
|
gst_amr_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
|
|
{
|
|
GstAmrParse *amrparse;
|
|
GstStructure *structure;
|
|
const gchar *name;
|
|
|
|
amrparse = GST_AMR_PARSE (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_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);
|
|
gst_amr_parse_set_src_caps (amrparse);
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_amr_parse_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_amr_parse_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_amr_parse_set_src_caps (amrparse);
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_amr_parse_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.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_amr_parse_handle_frame (GstBaseParse * parse,
|
|
GstBaseParseFrame * frame, gint * skipsize)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstMapInfo map;
|
|
gint fsize = 0, mode, dsize;
|
|
GstAmrParse *amrparse;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gboolean found = FALSE;
|
|
|
|
amrparse = GST_AMR_PARSE (parse);
|
|
buffer = frame->buffer;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
dsize = map.size;
|
|
|
|
GST_LOG ("buffer: %d bytes", dsize);
|
|
|
|
if (amrparse->need_header) {
|
|
if (dsize >= AMR_MIME_HEADER_SIZE &&
|
|
gst_amr_parse_parse_header (amrparse, map.data, skipsize)) {
|
|
amrparse->need_header = FALSE;
|
|
gst_base_parse_set_frame_rate (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. */
|
|
goto done;
|
|
}
|
|
|
|
*skipsize = 1;
|
|
/* Does this look like a possible frame header candidate? */
|
|
if ((map.data[0] & 0x83) == 0) {
|
|
/* Yep. Retrieve the frame size */
|
|
mode = (map.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) {
|
|
*skipsize = 0;
|
|
/* in sync, no further check */
|
|
if (!GST_BASE_PARSE_LOST_SYNC (parse)) {
|
|
found = TRUE;
|
|
} else if (dsize > fsize) {
|
|
/* enough data, check for next sync */
|
|
if ((map.data[fsize] & 0x83) == 0)
|
|
found = TRUE;
|
|
} else if (GST_BASE_PARSE_DRAINING (parse)) {
|
|
/* not enough, but draining, so ok */
|
|
found = TRUE;
|
|
}
|
|
}
|
|
}
|
|
|
|
done:
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
if (found && fsize <= map.size) {
|
|
ret = gst_base_parse_finish_frame (parse, frame, fsize);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_amr_parse_start:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "start" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
static gboolean
|
|
gst_amr_parse_start (GstBaseParse * parse)
|
|
{
|
|
GstAmrParse *amrparse;
|
|
|
|
amrparse = GST_AMR_PARSE (parse);
|
|
GST_DEBUG ("start");
|
|
amrparse->need_header = TRUE;
|
|
amrparse->header = 0;
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_amr_parse_stop:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "stop" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
static gboolean
|
|
gst_amr_parse_stop (GstBaseParse * parse)
|
|
{
|
|
GstAmrParse *amrparse;
|
|
|
|
amrparse = GST_AMR_PARSE (parse);
|
|
GST_DEBUG ("stop");
|
|
amrparse->need_header = TRUE;
|
|
amrparse->header = 0;
|
|
return TRUE;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_amr_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
|
|
{
|
|
GstCaps *peercaps, *templ;
|
|
GstCaps *res;
|
|
|
|
|
|
templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
|
|
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), filter);
|
|
|
|
if (peercaps) {
|
|
guint i, n;
|
|
|
|
/* Rename structure names */
|
|
peercaps = gst_caps_make_writable (peercaps);
|
|
n = gst_caps_get_size (peercaps);
|
|
for (i = 0; i < n; i++) {
|
|
GstStructure *s = gst_caps_get_structure (peercaps, i);
|
|
|
|
if (gst_structure_has_name (s, "audio/AMR"))
|
|
gst_structure_set_name (s, "audio/x-amr-nb-sh");
|
|
else
|
|
gst_structure_set_name (s, "audio/x-amr-wb-sh");
|
|
}
|
|
|
|
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (peercaps);
|
|
res = gst_caps_make_writable (res);
|
|
/* Append the template caps because we still want to accept
|
|
* caps without any fields in the case upstream does not
|
|
* know anything.
|
|
*/
|
|
gst_caps_append (res, templ);
|
|
} else {
|
|
res = templ;
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *intersection;
|
|
|
|
intersection =
|
|
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (res);
|
|
res = intersection;
|
|
}
|
|
|
|
return res;
|
|
}
|