gstreamer/ext/opus/gstopusparse.c

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/* GStreamer
* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Copyright (C) <2011-2012> Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
*
* 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-opusparse
* @title: opusparse
* @see_also: opusenc, opusdec
*
* This element parses OPUS packets.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=opusdata ! opusparse ! opusdec ! audioconvert ! audioresample ! alsasink
* ]| Decode and plays an unmuxed Opus file.
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
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#include <string.h>
#include <opus.h>
#include "gstopusheader.h"
#include "gstopusparse.h"
#include <gst/audio/audio.h>
#include <gst/pbutils/pbutils.h>
#include <gst/tag/tag.h>
GST_DEBUG_CATEGORY_STATIC (opusparse_debug);
#define GST_CAT_DEFAULT opusparse_debug
#define MAX_PAYLOAD_BYTES 1500
static GstStaticPadTemplate opus_parse_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, framed = (boolean) true")
);
static GstStaticPadTemplate opus_parse_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus")
);
static gboolean gst_opus_parse_start (GstBaseParse * parse);
static gboolean gst_opus_parse_stop (GstBaseParse * parse);
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static GstFlowReturn gst_opus_parse_handle_frame (GstBaseParse * base,
GstBaseParseFrame * frame, gint * skip);
static GstFlowReturn gst_opus_parse_parse_frame (GstBaseParse * base,
GstBaseParseFrame * frame);
static gboolean opusparse_element_init (GstPlugin * plugin);
G_DEFINE_TYPE (GstOpusParse, gst_opus_parse, GST_TYPE_BASE_PARSE);
GST_ELEMENT_REGISTER_DEFINE_CUSTOM (opusparse, opusparse_element_init);
static void
gst_opus_parse_class_init (GstOpusParseClass * klass)
{
GstBaseParseClass *bpclass;
GstElementClass *element_class;
bpclass = (GstBaseParseClass *) klass;
element_class = (GstElementClass *) klass;
bpclass->start = GST_DEBUG_FUNCPTR (gst_opus_parse_start);
bpclass->stop = GST_DEBUG_FUNCPTR (gst_opus_parse_stop);
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bpclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_parse_handle_frame);
gst_element_class_add_static_pad_template (element_class,
&opus_parse_src_factory);
gst_element_class_add_static_pad_template (element_class,
&opus_parse_sink_factory);
gst_element_class_set_static_metadata (element_class, "Opus audio parser",
"Codec/Parser/Audio", "parses opus audio streams",
"Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (opusparse_debug, "opusparse", 0,
"opus parsing element");
}
static void
gst_opus_parse_init (GstOpusParse * parse)
{
parse->header_sent = FALSE;
parse->got_headers = FALSE;
parse->pre_skip = 0;
}
static gboolean
gst_opus_parse_start (GstBaseParse * base)
{
GstOpusParse *parse = GST_OPUS_PARSE (base);
parse->header_sent = FALSE;
parse->got_headers = FALSE;
parse->pre_skip = 0;
parse->next_ts = 0;
return TRUE;
}
static gboolean
gst_opus_parse_stop (GstBaseParse * base)
{
GstOpusParse *parse = GST_OPUS_PARSE (base);
parse->header_sent = FALSE;
parse->got_headers = FALSE;
parse->pre_skip = 0;
return TRUE;
}
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static GstFlowReturn
gst_opus_parse_handle_frame (GstBaseParse * base,
GstBaseParseFrame * frame, gint * skip)
{
GstOpusParse *parse;
guint8 *data;
gsize size;
guint32 packet_size;
int ret = FALSE;
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const unsigned char *frames[48];
unsigned char toc;
short frame_sizes[48];
int payload_offset;
int packet_offset = 0;
gboolean is_header, is_idheader, is_commentheader;
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GstMapInfo map;
parse = GST_OPUS_PARSE (base);
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*skip = -1;
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
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GST_DEBUG_OBJECT (parse,
"Checking for frame, %" G_GSIZE_FORMAT " bytes in buffer", size);
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/* check for headers */
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is_idheader = gst_opus_header_is_id_header (frame->buffer);
is_commentheader = gst_opus_header_is_comment_header (frame->buffer);
is_header = is_idheader || is_commentheader;
if (!is_header) {
int nframes;
/* Next, check if there's an Opus packet there */
nframes =
opus_packet_parse (data, size, &toc, frames, frame_sizes,
&payload_offset);
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if (nframes < 0) {
/* Then, check for the test vector framing */
GST_DEBUG_OBJECT (parse,
"No Opus packet found, trying test vector framing");
if (size < 4) {
GST_DEBUG_OBJECT (parse, "Too small");
goto beach;
}
packet_size = GST_READ_UINT32_BE (data);
GST_DEBUG_OBJECT (parse, "Packet size: %u bytes", packet_size);
if (packet_size > MAX_PAYLOAD_BYTES) {
GST_DEBUG_OBJECT (parse, "Too large");
goto beach;
}
if (packet_size > size - 4) {
GST_DEBUG_OBJECT (parse, "Truncated");
goto beach;
}
nframes =
opus_packet_parse (data + 8, packet_size, &toc, frames, frame_sizes,
&payload_offset);
if (nframes < 0) {
GST_DEBUG_OBJECT (parse, "No test vector framing either");
goto beach;
}
packet_offset = 8;
/* for ad hoc framing, heed the framing, so we eat any padding */
payload_offset = packet_size;
} else {
/* Add up all the frame sizes found */
int f;
for (f = 0; f < nframes; ++f)
payload_offset += frame_sizes[f];
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}
}
if (is_header) {
*skip = 0;
} else {
*skip = packet_offset;
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size = payload_offset;
}
GST_DEBUG_OBJECT (parse,
"Got Opus packet at offset %d, %" G_GSIZE_FORMAT " bytes", *skip, size);
ret = TRUE;
beach:
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gst_buffer_unmap (frame->buffer, &map);
/* convert old style result to new one */
if (!ret) {
if (*skip < 0)
*skip = 1;
return GST_FLOW_OK;
}
/* always skip first if needed */
if (*skip > 0)
return GST_FLOW_OK;
/* normalize again */
if (*skip < 0)
*skip = 0;
/* not enough */
if (size > map.size)
return GST_FLOW_OK;
/* FIXME some day ... should not mess with buffer itself */
if (!parse->got_headers) {
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gst_buffer_replace (&frame->buffer,
gst_buffer_copy_region (frame->buffer, GST_BUFFER_COPY_ALL, 0, size));
gst_buffer_unref (frame->buffer);
}
ret = gst_opus_parse_parse_frame (base, frame);
if (ret == GST_BASE_PARSE_FLOW_DROPPED) {
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_DROP;
ret = GST_FLOW_OK;
}
if (ret == GST_FLOW_OK)
ret = gst_base_parse_finish_frame (base, frame, size);
return ret;
}
/* Adapted copy of the one in gstoggstream.c... */
static guint64
packet_duration_opus (const guint8 * data, size_t len)
{
static const guint64 durations[32] = {
10000, 20000, 40000, 60000, /* Silk NB */
10000, 20000, 40000, 60000, /* Silk MB */
10000, 20000, 40000, 60000, /* Silk WB */
10000, 20000, /* Hybrid SWB */
10000, 20000, /* Hybrid FB */
2500, 5000, 10000, 20000, /* CELT NB */
2500, 5000, 10000, 20000, /* CELT NB */
2500, 5000, 10000, 20000, /* CELT NB */
2500, 5000, 10000, 20000, /* CELT NB */
};
gint64 duration;
gint64 frame_duration;
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gint nframes = 0;
guint8 toc;
if (len < 1)
return 0;
toc = data[0];
frame_duration = durations[toc >> 3] * 1000;
switch (toc & 3) {
case 0:
nframes = 1;
break;
case 1:
nframes = 2;
break;
case 2:
nframes = 2;
break;
case 3:
if (len < 2) {
GST_WARNING ("Code 3 Opus packet has less than 2 bytes");
return 0;
}
nframes = data[1] & 63;
break;
}
duration = nframes * frame_duration;
if (duration > 120 * GST_MSECOND) {
GST_WARNING ("Opus packet duration > 120 ms, invalid");
return 0;
}
GST_LOG ("Opus packet: frame size %.1f ms, %d frames, duration %.1f ms",
frame_duration / 1000000.f, nframes, duration / 1000000.f);
return duration;
}
static GstFlowReturn
gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
{
guint64 duration;
GstOpusParse *parse;
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gboolean is_idheader, is_commentheader;
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GstMapInfo map;
GstAudioClippingMeta *cmeta =
gst_buffer_get_audio_clipping_meta (frame->buffer);
parse = GST_OPUS_PARSE (base);
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
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is_idheader = gst_opus_header_is_id_header (frame->buffer);
is_commentheader = gst_opus_header_is_comment_header (frame->buffer);
if (!parse->got_headers || !parse->header_sent) {
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GstCaps *caps;
/* Opus streams can decode to 1 or 2 channels, so use the header
value if we have one, or 2 otherwise */
if (is_idheader) {
gst_buffer_replace (&parse->id_header, frame->buffer);
GST_DEBUG_OBJECT (parse, "Found ID header, keeping");
return GST_BASE_PARSE_FLOW_DROPPED;
} else if (is_commentheader) {
gst_buffer_replace (&parse->comment_header, frame->buffer);
GST_DEBUG_OBJECT (parse, "Found comment header, keeping");
return GST_BASE_PARSE_FLOW_DROPPED;
}
parse->got_headers = TRUE;
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if (cmeta && cmeta->start) {
parse->pre_skip += cmeta->start;
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
duration = packet_duration_opus (map.data, map.size);
gst_buffer_unmap (frame->buffer, &map);
/* Queue frame for later once we know all initial padding */
if (duration == cmeta->start) {
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_QUEUE;
}
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}
if (!(frame->flags & GST_BASE_PARSE_FRAME_FLAG_QUEUE)) {
GstCaps *sink_caps;
guint32 sample_rate = 48000;
guint8 n_channels, n_streams, n_stereo_streams, channel_mapping_family;
guint8 channel_mapping[256];
GstBuffer *id_header;
guint16 pre_skip = 0;
gint16 gain = 0;
if (parse->id_header) {
gst_buffer_map (parse->id_header, &map, GST_MAP_READWRITE);
pre_skip = GST_READ_UINT16_LE (map.data + 10);
gain = GST_READ_UINT16_LE (map.data + 16);
gst_buffer_unmap (parse->id_header, &map);
}
sink_caps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (parse));
if (!sink_caps
|| !gst_codec_utils_opus_parse_caps (sink_caps, &sample_rate,
&n_channels, &channel_mapping_family, &n_streams,
&n_stereo_streams, channel_mapping)) {
GST_INFO_OBJECT (parse,
"No headers and no caps, blindly setting up canonical stereo");
n_channels = 2;
n_streams = 1;
n_stereo_streams = 1;
channel_mapping_family = 0;
channel_mapping[0] = 0;
channel_mapping[1] = 1;
}
if (sink_caps)
gst_caps_unref (sink_caps);
id_header =
gst_codec_utils_opus_create_header (sample_rate, n_channels,
channel_mapping_family, n_streams, n_stereo_streams,
channel_mapping, pre_skip, gain);
caps = gst_codec_utils_opus_create_caps_from_header (id_header, NULL);
gst_buffer_unref (id_header);
gst_buffer_replace (&parse->id_header, NULL);
gst_buffer_replace (&parse->comment_header, NULL);
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gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
parse->header_sent = TRUE;
}
}
GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;
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gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
duration = packet_duration_opus (map.data, map.size);
gst_buffer_unmap (frame->buffer, &map);
parse->next_ts += duration;
GST_BUFFER_DURATION (frame->buffer) = duration;
GST_BUFFER_OFFSET_END (frame->buffer) =
gst_util_uint64_scale (parse->next_ts, 48000, GST_SECOND);
GST_BUFFER_OFFSET (frame->buffer) = parse->next_ts;
return GST_FLOW_OK;
}
static gboolean
opusparse_element_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "opusparse", GST_RANK_NONE,
GST_TYPE_OPUS_PARSE))
return FALSE;
gst_tag_register_musicbrainz_tags ();
return TRUE;
}