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c145798876
gst_caps_new_simple gets wrong types for rate and channel which may lead to a crash. As 64-bit values for rate, depth, format, channels does not make much sense and since any other functionality in gstreamer expects G_TYPE_INT for channels and rate, we should stick to that Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1576>
326 lines
10 KiB
C
326 lines
10 KiB
C
/*
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* GStreamer AVTP Plugin
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* Copyright (C) 2019 Intel Corporation
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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 Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later
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* 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser 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 Street, 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-avtpaafdepay
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* @see_also: avtpaafpay
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*
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* Extract raw audio from AVTPDUs according to IEEE 1722-2016. For detailed
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* information see https://standards.ieee.org/standard/1722-2016.html.
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch-1.0 avtpsrc ! avtpaafdepay ! autoaudiosink
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* ]| This example pipeline will depayload AVTPDUs. Refer to the avtpaafpay
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* example to create the AVTP stream.
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* </refsect2>
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*/
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#include <avtp.h>
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#include <avtp_aaf.h>
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#include <gst/audio/audio-format.h>
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#include "gstavtpaafdepay.h"
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GST_DEBUG_CATEGORY_STATIC (avtpaafdepay_debug);
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#define GST_CAT_DEFAULT (avtpaafdepay_debug)
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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/x-raw, "
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"format = (string) { S16BE, S24BE, S32BE, F32BE }, "
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"rate = (int) { 8000, 16000, 24000, 32000, 44100, 48000, 88200, 96000, 176400, 192000 }, "
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"channels = " GST_AUDIO_CHANNELS_RANGE ", "
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"layout = (string) interleaved")
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);
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G_DEFINE_TYPE (GstAvtpAafDepay, gst_avtp_aaf_depay,
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GST_TYPE_AVTP_BASE_DEPAYLOAD);
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static GstFlowReturn gst_avtp_aaf_depay_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static void
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gst_avtp_aaf_depay_class_init (GstAvtpAafDepayClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAvtpBaseDepayloadClass *avtpbasedepayload_class =
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GST_AVTP_BASE_DEPAYLOAD_CLASS (klass);
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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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"AVTP Audio Format (AAF) depayloader",
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"Codec/Depayloader/Network/AVTP",
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"Extracts raw audio from AAF AVTPDUs",
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"Andre Guedes <andre.guedes@intel.com>");
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avtpbasedepayload_class->chain = GST_DEBUG_FUNCPTR (gst_avtp_aaf_depay_chain);
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GST_DEBUG_CATEGORY_INIT (avtpaafdepay_debug, "avtpaafdepay", 0,
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"AAF AVTP Depayloader");
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}
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static void
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gst_avtp_aaf_depay_init (GstAvtpAafDepay * avtpaafdepay)
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{
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avtpaafdepay->channels = 0;
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avtpaafdepay->depth = 0;
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avtpaafdepay->rate = 0;
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avtpaafdepay->format = 0;
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}
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static const gchar *
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avtp_to_gst_format (int avtp_format)
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{
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GstAudioFormat gst_format;
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switch (avtp_format) {
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case AVTP_AAF_FORMAT_INT_16BIT:
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gst_format = GST_AUDIO_FORMAT_S16BE;
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break;
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case AVTP_AAF_FORMAT_INT_24BIT:
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gst_format = GST_AUDIO_FORMAT_S24BE;
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break;
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case AVTP_AAF_FORMAT_INT_32BIT:
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gst_format = GST_AUDIO_FORMAT_S32BE;
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break;
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case AVTP_AAF_FORMAT_FLOAT_32BIT:
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gst_format = GST_AUDIO_FORMAT_F32BE;
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break;
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default:
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gst_format = GST_AUDIO_FORMAT_UNKNOWN;
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break;
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}
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return gst_audio_format_to_string (gst_format);
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}
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static gint
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avtp_to_gst_rate (int rate)
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{
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switch (rate) {
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case AVTP_AAF_PCM_NSR_8KHZ:
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return 8000;
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case AVTP_AAF_PCM_NSR_16KHZ:
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return 16000;
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case AVTP_AAF_PCM_NSR_24KHZ:
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return 24000;
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case AVTP_AAF_PCM_NSR_32KHZ:
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return 32000;
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case AVTP_AAF_PCM_NSR_44_1KHZ:
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return 44100;
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case AVTP_AAF_PCM_NSR_48KHZ:
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return 48000;
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case AVTP_AAF_PCM_NSR_88_2KHZ:
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return 88200;
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case AVTP_AAF_PCM_NSR_96KHZ:
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return 96000;
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case AVTP_AAF_PCM_NSR_176_4KHZ:
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return 176400;
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case AVTP_AAF_PCM_NSR_192KHZ:
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return 192000;
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default:
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return 0;
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}
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}
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static gboolean
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gst_avtp_aaf_depay_push_caps_event (GstAvtpAafDepay * avtpaafdepay,
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gint rate, gint depth, gint format, gint channels)
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{
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GstCaps *caps;
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GstEvent *event;
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GstAvtpBaseDepayload *avtpbasedepayload =
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GST_AVTP_BASE_DEPAYLOAD (avtpaafdepay);
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, avtp_to_gst_format (format),
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"rate", G_TYPE_INT, avtp_to_gst_rate (rate),
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"channels", G_TYPE_INT, channels,
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"layout", G_TYPE_STRING, "interleaved", NULL);
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event = gst_event_new_caps (caps);
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if (!gst_pad_push_event (avtpbasedepayload->srcpad, event)) {
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GST_ERROR_OBJECT (avtpaafdepay, "Failed to push CAPS event");
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gst_caps_unref (caps);
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return FALSE;
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}
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GST_DEBUG_OBJECT (avtpaafdepay, "CAPS event pushed %" GST_PTR_FORMAT, caps);
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avtpaafdepay->rate = rate;
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avtpaafdepay->depth = depth;
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avtpaafdepay->format = format;
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avtpaafdepay->channels = channels;
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gst_caps_unref (caps);
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return TRUE;
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}
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static gboolean
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gst_avtp_aaf_depay_are_audio_features_valid (GstAvtpAafDepay * avtpaafdepay,
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guint64 rate, guint64 depth, guint64 format, guint64 channels)
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{
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if (G_UNLIKELY (rate != avtpaafdepay->rate)) {
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GST_INFO_OBJECT (avtpaafdepay, "Rate doesn't match, disarding buffer");
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return FALSE;
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}
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if (G_UNLIKELY (depth != avtpaafdepay->depth)) {
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GST_INFO_OBJECT (avtpaafdepay, "Bit depth doesn't match, disarding buffer");
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return FALSE;
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}
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if (G_UNLIKELY (format != avtpaafdepay->format)) {
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GST_INFO_OBJECT (avtpaafdepay,
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"Sample format doesn't match, disarding buffer");
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return FALSE;
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}
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if (G_UNLIKELY (channels != avtpaafdepay->channels)) {
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GST_INFO_OBJECT (avtpaafdepay,
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"Number of channels doesn't match, disarding buffer");
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return FALSE;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_avtp_aaf_depay_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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int res;
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GstMapInfo info;
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guint32 subtype, version;
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GstClockTime ptime;
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GstBuffer *subbuffer;
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struct avtp_stream_pdu *pdu;
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guint64 channels, depth, rate, format, tstamp, seqnum, streamid,
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streamid_valid, data_len;
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GstAvtpBaseDepayload *avtpbasedepayload = GST_AVTP_BASE_DEPAYLOAD (parent);
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GstAvtpAafDepay *avtpaafdepay = GST_AVTP_AAF_DEPAY (avtpbasedepayload);
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if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (avtpaafdepay, RESOURCE, READ, ("Failed to map memory"),
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(NULL));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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if (info.size < sizeof (struct avtp_stream_pdu)) {
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GST_DEBUG_OBJECT (avtpaafdepay, "Malformed AVTPDU, discarding it");
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gst_buffer_unmap (buffer, &info);
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goto discard;
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}
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pdu = (struct avtp_stream_pdu *) info.data;
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_NSR, &rate);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_FORMAT, &format);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_SEQ_NUM, &seqnum);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_BIT_DEPTH, &depth);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_TIMESTAMP, &tstamp);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_SV, &streamid_valid);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_STREAM_ID, &streamid);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_CHAN_PER_FRAME, &channels);
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g_assert (res == 0);
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res = avtp_aaf_pdu_get (pdu, AVTP_AAF_FIELD_STREAM_DATA_LEN, &data_len);
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g_assert (res == 0);
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res = avtp_pdu_get ((struct avtp_common_pdu *) pdu, AVTP_FIELD_SUBTYPE,
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&subtype);
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g_assert (res == 0);
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res = avtp_pdu_get ((struct avtp_common_pdu *) pdu, AVTP_FIELD_VERSION,
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&version);
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g_assert (res == 0);
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gst_buffer_unmap (buffer, &info);
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if (subtype != AVTP_SUBTYPE_AAF) {
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GST_DEBUG_OBJECT (avtpaafdepay, "Subtype doesn't match, discarding buffer");
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goto discard;
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}
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if (version != 0) {
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GST_DEBUG_OBJECT (avtpaafdepay, "Version doesn't match, discarding buffer");
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goto discard;
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}
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if (streamid_valid != 1 || streamid != avtpbasedepayload->streamid) {
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GST_DEBUG_OBJECT (avtpaafdepay, "Invalid StreamID, discarding buffer");
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goto discard;
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}
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if (gst_buffer_get_size (buffer) < sizeof (*pdu) + data_len) {
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GST_DEBUG_OBJECT (avtpaafdepay, "Incomplete AVTPDU, discarding buffer");
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goto discard;
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}
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if (G_UNLIKELY (!gst_pad_has_current_caps (avtpbasedepayload->srcpad))) {
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if (!gst_avtp_aaf_depay_push_caps_event (avtpaafdepay, rate, depth, format,
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channels)) {
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gst_buffer_unref (buffer);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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if (!gst_avtp_base_depayload_push_segment_event (avtpbasedepayload, tstamp)) {
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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avtpbasedepayload->seqnum = seqnum;
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}
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if (G_UNLIKELY (!gst_avtp_aaf_depay_are_audio_features_valid (avtpaafdepay,
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rate, depth, format, channels)))
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goto discard;
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if (seqnum != avtpbasedepayload->seqnum) {
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GST_INFO_OBJECT (avtpaafdepay, "Sequence number mismatch: expected %u"
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" received %" G_GUINT64_FORMAT, avtpbasedepayload->seqnum, seqnum);
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avtpbasedepayload->seqnum = seqnum;
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}
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avtpbasedepayload->seqnum++;
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ptime = gst_avtp_base_depayload_tstamp_to_ptime (avtpbasedepayload, tstamp,
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avtpbasedepayload->prev_ptime);
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subbuffer = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
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sizeof (struct avtp_stream_pdu), data_len);
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GST_BUFFER_PTS (subbuffer) = ptime;
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GST_BUFFER_DTS (subbuffer) = ptime;
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avtpbasedepayload->prev_ptime = ptime;
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gst_buffer_unref (buffer);
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return gst_pad_push (avtpbasedepayload->srcpad, subbuffer);
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discard:
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
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}
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gboolean
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gst_avtp_aaf_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "avtpaafdepay", GST_RANK_NONE,
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GST_TYPE_AVTP_AAF_DEPAY);
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}
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