mirror of
https://github.com/irungentoo/toxcore.git
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This can be used by fuzzers to make RNG-driven code deterministic (i.e. based on the fuzzer input).
872 lines
31 KiB
C
872 lines
31 KiB
C
/* SPDX-License-Identifier: GPL-3.0-or-later
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* Copyright © 2016-2018 The TokTok team.
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* Copyright © 2013-2015 Tox project.
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*/
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#include "rtp.h"
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#include <assert.h>
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#include <errno.h>
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#include <stdlib.h>
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#include <string.h>
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#include "bwcontroller.h"
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#include "../toxcore/Messenger.h"
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#include "../toxcore/ccompat.h"
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#include "../toxcore/logger.h"
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#include "../toxcore/mono_time.h"
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#include "../toxcore/util.h"
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/**
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* The number of milliseconds we want to keep a keyframe in the buffer for,
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* even though there are no free slots for incoming frames.
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*/
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#define VIDEO_KEEP_KEYFRAME_IN_BUFFER_FOR_MS 15
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/**
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* return -1 on failure, 0 on success
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*
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*/
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static int rtp_send_custom_lossy_packet(Tox *tox, int32_t friendnumber, const uint8_t *data, uint32_t length)
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{
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Tox_Err_Friend_Custom_Packet error;
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tox_friend_send_lossy_packet(tox, friendnumber, data, (size_t)length, &error);
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if (error == TOX_ERR_FRIEND_CUSTOM_PACKET_OK) {
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return 0;
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}
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return -1;
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}
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// allocate_len is NOT including header!
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static struct RTPMessage *new_message(const struct RTPHeader *header, size_t allocate_len, const uint8_t *data,
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uint16_t data_length)
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{
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assert(allocate_len >= data_length);
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struct RTPMessage *msg = (struct RTPMessage *)calloc(1, sizeof(struct RTPMessage) + allocate_len);
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if (msg == nullptr) {
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return nullptr;
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}
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msg->len = data_length; // result without header
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msg->header = *header;
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memcpy(msg->data, data, msg->len);
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return msg;
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}
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/**
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* Instruct the caller to clear slot 0.
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*/
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#define GET_SLOT_RESULT_DROP_OLDEST_SLOT (-1)
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/**
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* Instruct the caller to drop the incoming packet.
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*/
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#define GET_SLOT_RESULT_DROP_INCOMING (-2)
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/**
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* Find the next free slot in work_buffer for the incoming data packet.
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*
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* - If the data packet belongs to a frame that's already in the work_buffer then
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* use that slot.
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* - If there is no free slot return GET_SLOT_RESULT_DROP_OLDEST_SLOT.
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* - If the data packet is too old return GET_SLOT_RESULT_DROP_INCOMING.
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*
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* If there is a keyframe being assembled in slot 0, keep it a bit longer and
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* do not kick it out right away if all slots are full instead kick out the new
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* incoming interframe.
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*/
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static int8_t get_slot(const Logger *log, struct RTPWorkBufferList *wkbl, bool is_keyframe,
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const struct RTPHeader *header, bool is_multipart)
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{
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if (is_multipart) {
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// This RTP message is part of a multipart frame, so we try to find an
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// existing slot with the previous parts of the frame in it.
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for (uint8_t i = 0; i < wkbl->next_free_entry; ++i) {
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const struct RTPWorkBuffer *slot = &wkbl->work_buffer[i];
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if ((slot->buf->header.sequnum == header->sequnum) && (slot->buf->header.timestamp == header->timestamp)) {
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// Sequence number and timestamp match, so this slot belongs to
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// the same frame.
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//
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// In reality, these will almost certainly either both match or
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// both not match. Only if somehow there were 65535 frames
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// between, the timestamp will matter.
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return i;
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}
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}
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}
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// The message may or may not be part of a multipart frame.
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//
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// If it is part of a multipart frame, then this is an entirely new frame
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// for which we did not have a slot *or* the frame is so old that its slot
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// has been evicted by now.
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//
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// |----------- time ----------->
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// _________________
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// slot 0 | |
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// -----------------
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// _________________
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// slot 1 | |
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// -----------------
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// ____________
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// slot 2 | | -> frame too old, drop
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// ------------
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//
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//
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//
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// |----------- time ----------->
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// _________________
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// slot 0 | |
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// -----------------
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// _________________
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// slot 1 | |
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// -----------------
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// ____________
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// slot 2 | | -> ok, start filling in a new slot
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// ------------
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// If there is a free slot:
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if (wkbl->next_free_entry < USED_RTP_WORKBUFFER_COUNT) {
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// If there is at least one filled slot:
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if (wkbl->next_free_entry > 0) {
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// Get the most recently filled slot.
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const struct RTPWorkBuffer *slot = &wkbl->work_buffer[wkbl->next_free_entry - 1];
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// If the incoming packet is older than our newest slot, drop it.
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// This is the first situation in the above diagram.
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if (slot->buf->header.timestamp > header->timestamp) {
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LOGGER_DEBUG(log, "workbuffer:2:timestamp too old");
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return GET_SLOT_RESULT_DROP_INCOMING;
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}
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}
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// Not all slots are filled, and the packet is newer than our most
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// recent slot, so it's a new frame we want to start assembling. This is
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// the second situation in the above diagram.
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return wkbl->next_free_entry;
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}
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// If the incoming frame is a key frame, then stop assembling the oldest
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// slot, regardless of whether there was a keyframe in that or not.
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if (is_keyframe) {
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return GET_SLOT_RESULT_DROP_OLDEST_SLOT;
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}
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// The incoming slot is not a key frame, so we look at slot 0 to see what to
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// do next.
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const struct RTPWorkBuffer *slot = &wkbl->work_buffer[0];
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// The incoming frame is not a key frame, but the existing slot 0 is also
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// not a keyframe, so we stop assembling the existing frame and make space
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// for the new one.
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if (!slot->is_keyframe) {
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return GET_SLOT_RESULT_DROP_OLDEST_SLOT;
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}
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// If this key frame is fully received, we also stop assembling and clear
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// slot 0. This also means sending the frame to the decoder.
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if (slot->received_len == slot->buf->header.data_length_full) {
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return GET_SLOT_RESULT_DROP_OLDEST_SLOT;
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}
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// This is a key frame, not fully received yet, but it's already much older
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// than the incoming frame, so we stop assembling it and send whatever part
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// we did receive to the decoder.
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if (slot->buf->header.timestamp + VIDEO_KEEP_KEYFRAME_IN_BUFFER_FOR_MS <= header->timestamp) {
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return GET_SLOT_RESULT_DROP_OLDEST_SLOT;
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}
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// This is a key frame, it's not too old yet, so we keep it in its slot for
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// a little longer.
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LOGGER_INFO(log, "keep KEYFRAME in workbuffer");
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return GET_SLOT_RESULT_DROP_INCOMING;
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}
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/**
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* Returns an assembled frame (as much data as we currently have for this frame,
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* some pieces may be missing)
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*
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* If there are no frames ready, we return NULL. If this function returns
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* non-NULL, it transfers ownership of the message to the caller, i.e. the
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* caller is responsible for storing it elsewhere or calling `free()`.
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*/
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static struct RTPMessage *process_frame(const Logger *log, struct RTPWorkBufferList *wkbl, uint8_t slot_id)
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{
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assert(wkbl->next_free_entry >= 0);
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if (wkbl->next_free_entry == 0) {
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// There are no frames in any slot.
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return nullptr;
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}
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// Slot 0 contains a key frame, slot_id points at an interframe that is
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// relative to that key frame, so we don't use it yet.
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if (wkbl->work_buffer[0].is_keyframe && slot_id != 0) {
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LOGGER_DEBUG(log, "process_frame:KEYFRAME waiting in slot 0");
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return nullptr;
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}
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// Either slot_id is 0 and slot 0 is a key frame, or there is no key frame
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// in slot 0 (and slot_id is anything).
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struct RTPWorkBuffer *const slot = &wkbl->work_buffer[slot_id];
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// Move ownership of the frame out of the slot into m_new.
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struct RTPMessage *const m_new = slot->buf;
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slot->buf = nullptr;
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assert(wkbl->next_free_entry >= 1 && wkbl->next_free_entry <= USED_RTP_WORKBUFFER_COUNT);
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if (slot_id != wkbl->next_free_entry - 1) {
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// The slot is not the last slot, so we created a gap. We move all the
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// entries after it one step up.
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for (uint8_t i = slot_id; i < wkbl->next_free_entry - 1; ++i) {
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// Move entry (i+1) into entry (i).
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wkbl->work_buffer[i] = wkbl->work_buffer[i + 1];
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}
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}
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// We now have a free entry at the end of the array.
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--wkbl->next_free_entry;
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// Clear the newly freed entry.
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const struct RTPWorkBuffer empty = {0};
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wkbl->work_buffer[wkbl->next_free_entry] = empty;
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// Move ownership of the frame to the caller.
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return m_new;
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}
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/**
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* @param log A logger.
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* @param wkbl The list of in-progress frames, i.e. all the slots.
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* @param slot_id The slot we want to fill the data into.
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* @param is_keyframe Whether the data is part of a key frame.
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* @param header The RTP header from the incoming packet.
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* @param incoming_data The pure payload without header.
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* @param incoming_data_length The length in bytes of the incoming data payload.
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*/
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static bool fill_data_into_slot(const Logger *log, struct RTPWorkBufferList *wkbl, const uint8_t slot_id,
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bool is_keyframe, const struct RTPHeader *header,
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const uint8_t *incoming_data, uint16_t incoming_data_length)
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{
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// We're either filling the data into an existing slot, or in a new one that
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// is the next free entry.
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assert(slot_id <= wkbl->next_free_entry);
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struct RTPWorkBuffer *const slot = &wkbl->work_buffer[slot_id];
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assert(header != nullptr);
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assert(is_keyframe == (bool)((header->flags & RTP_KEY_FRAME) != 0));
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if (slot->received_len == 0) {
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assert(slot->buf == nullptr);
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// No data for this slot has been received, yet, so we create a new
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// message for it with enough memory for the entire frame.
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struct RTPMessage *msg = (struct RTPMessage *)calloc(1, sizeof(struct RTPMessage) + header->data_length_full);
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if (msg == nullptr) {
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LOGGER_ERROR(log, "Out of memory while trying to allocate for frame of size %u",
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(unsigned)header->data_length_full);
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// Out of memory: throw away the incoming data.
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return false;
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}
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// Unused in the new video receiving code, as it's 16 bit and can't hold
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// the full length of large frames. Instead, we use slot->received_len.
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msg->len = 0;
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msg->header = *header;
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slot->buf = msg;
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slot->is_keyframe = is_keyframe;
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slot->received_len = 0;
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assert(wkbl->next_free_entry < USED_RTP_WORKBUFFER_COUNT);
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++wkbl->next_free_entry;
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}
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// We already checked this when we received the packet, but we rely on it
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// here, so assert again.
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assert(header->offset_full < header->data_length_full);
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// Copy the incoming chunk of data into the correct position in the full
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// frame data array.
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memcpy(
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slot->buf->data + header->offset_full,
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incoming_data,
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incoming_data_length
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);
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// Update the total received length of this slot.
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slot->received_len += incoming_data_length;
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// Update received length also in the header of the message, for later use.
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slot->buf->header.received_length_full = slot->received_len;
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return slot->received_len == header->data_length_full;
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}
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static void update_bwc_values(const Logger *log, RTPSession *session, const struct RTPMessage *msg)
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{
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if (session->first_packets_counter < DISMISS_FIRST_LOST_VIDEO_PACKET_COUNT) {
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++session->first_packets_counter;
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} else {
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const uint32_t data_length_full = msg->header.data_length_full; // without header
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const uint32_t received_length_full = msg->header.received_length_full; // without header
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bwc_add_recv(session->bwc, data_length_full);
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if (received_length_full < data_length_full) {
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LOGGER_DEBUG(log, "BWC: full length=%u received length=%d", data_length_full, received_length_full);
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bwc_add_lost(session->bwc, data_length_full - received_length_full);
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}
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}
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}
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/**
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* Handle a single RTP video packet.
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*
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* The packet may or may not be part of a multipart frame. This function will
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* find out and handle it appropriately.
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*
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* @param session The current RTP session with:
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* <code>
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* session->mcb == vc_queue_message() // this function is called from here
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* session->mp == struct RTPMessage *
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* session->cs == call->video.second // == VCSession created by vc_new() call
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* </code>
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* @param header The RTP header deserialised from the packet.
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* @param incoming_data The packet data *not* header, i.e. this is the actual
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* payload.
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* @param incoming_data_length The packet length *not* including header, i.e.
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* this is the actual payload length.
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* @param log A logger.
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*
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* @retval -1 on error.
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* @retval 0 on success.
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*/
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static int handle_video_packet(RTPSession *session, const struct RTPHeader *header,
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const uint8_t *incoming_data, uint16_t incoming_data_length, const Logger *log)
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{
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// Full frame length in bytes. The frame may be split into multiple packets,
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// but this value is the complete assembled frame size.
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const uint32_t full_frame_length = header->data_length_full;
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// Current offset in the frame. If this is the first packet of a multipart
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// frame or it's not a multipart frame, then this value is 0.
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const uint32_t offset = header->offset_full; // without header
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// The sender tells us whether this is a key frame.
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const bool is_keyframe = (header->flags & RTP_KEY_FRAME) != 0;
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LOGGER_DEBUG(log, "-- handle_video_packet -- full lens=%u len=%u offset=%u is_keyframe=%s",
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(unsigned)incoming_data_length, (unsigned)full_frame_length, (unsigned)offset, is_keyframe ? "K" : ".");
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LOGGER_DEBUG(log, "wkbl->next_free_entry:003=%d", session->work_buffer_list->next_free_entry);
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const bool is_multipart = full_frame_length != incoming_data_length;
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/* The message was sent in single part */
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int8_t slot_id = get_slot(log, session->work_buffer_list, is_keyframe, header, is_multipart);
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LOGGER_DEBUG(log, "slot num=%d", slot_id);
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// get_slot told us to drop the packet, so we ignore it.
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if (slot_id == GET_SLOT_RESULT_DROP_INCOMING) {
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return -1;
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}
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// get_slot said there is no free slot.
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if (slot_id == GET_SLOT_RESULT_DROP_OLDEST_SLOT) {
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LOGGER_DEBUG(log, "there was no free slot, so we process the oldest frame");
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// We now own the frame.
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struct RTPMessage *m_new = process_frame(log, session->work_buffer_list, 0);
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// The process_frame function returns NULL if there is no slot 0, i.e.
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// the work buffer list is completely empty. It can't be empty, because
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// get_slot just told us it's full, so process_frame must return non-null.
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assert(m_new != nullptr);
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LOGGER_DEBUG(log, "-- handle_video_packet -- CALLBACK-001a b0=%d b1=%d", (int)m_new->data[0], (int)m_new->data[1]);
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update_bwc_values(log, session, m_new);
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// Pass ownership of m_new to the callback.
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session->mcb(session->m->mono_time, session->cs, m_new);
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// Now we no longer own m_new.
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m_new = nullptr;
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// Now we must have a free slot, so we either get that slot, i.e. >= 0,
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// or get told to drop the incoming packet if it's too old.
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slot_id = get_slot(log, session->work_buffer_list, is_keyframe, header, /* is_multipart */false);
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if (slot_id == GET_SLOT_RESULT_DROP_INCOMING) {
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// The incoming frame is too old, so we drop it.
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return -1;
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}
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}
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// We must have a valid slot here.
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assert(slot_id >= 0);
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LOGGER_DEBUG(log, "fill_data_into_slot.1");
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// fill in this part into the slot buffer at the correct offset
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if (!fill_data_into_slot(
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log,
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session->work_buffer_list,
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slot_id,
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is_keyframe,
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header,
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incoming_data,
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incoming_data_length)) {
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// Memory allocation failed. Return error.
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return -1;
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}
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struct RTPMessage *m_new = process_frame(log, session->work_buffer_list, slot_id);
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if (m_new != nullptr) {
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LOGGER_DEBUG(log, "-- handle_video_packet -- CALLBACK-003a b0=%d b1=%d", (int)m_new->data[0], (int)m_new->data[1]);
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update_bwc_values(log, session, m_new);
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session->mcb(session->m->mono_time, session->cs, m_new);
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m_new = nullptr;
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}
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return 0;
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}
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/**
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* @retval -1 on error.
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* @retval 0 on success.
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*/
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static int handle_rtp_packet(Messenger *m, uint32_t friendnumber, const uint8_t *data, uint16_t length, void *object)
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{
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RTPSession *session = (RTPSession *)object;
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if (session == nullptr || length < RTP_HEADER_SIZE + 1) {
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LOGGER_WARNING(m->log, "No session or invalid length of received buffer!");
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return -1;
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}
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// Get the packet type.
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const uint8_t packet_type = data[0];
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++data;
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--length;
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// Unpack the header.
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struct RTPHeader header;
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rtp_header_unpack(data, &header);
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if (header.pt != packet_type % 128) {
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LOGGER_WARNING(m->log, "RTPHeader packet type and Tox protocol packet type did not agree: %d != %d",
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header.pt, packet_type % 128);
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return -1;
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}
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|
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if (header.pt != session->payload_type % 128) {
|
|
LOGGER_WARNING(m->log, "RTPHeader packet type does not match this session's payload type: %d != %d",
|
|
header.pt, session->payload_type % 128);
|
|
return -1;
|
|
}
|
|
|
|
if ((header.flags & RTP_LARGE_FRAME) != 0 && header.offset_full >= header.data_length_full) {
|
|
LOGGER_ERROR(m->log, "Invalid video packet: frame offset (%u) >= full frame length (%u)",
|
|
(unsigned)header.offset_full, (unsigned)header.data_length_full);
|
|
return -1;
|
|
}
|
|
|
|
if (header.offset_lower >= header.data_length_lower) {
|
|
LOGGER_ERROR(m->log, "Invalid old protocol video packet: frame offset (%u) >= full frame length (%u)",
|
|
(unsigned)header.offset_lower, (unsigned)header.data_length_lower);
|
|
return -1;
|
|
}
|
|
|
|
LOGGER_DEBUG(m->log, "header.pt %d, video %d", (uint8_t)header.pt, RTP_TYPE_VIDEO % 128);
|
|
|
|
// The sender uses the new large-frame capable protocol and is sending a
|
|
// video packet.
|
|
if ((header.flags & RTP_LARGE_FRAME) != 0 && header.pt == (RTP_TYPE_VIDEO % 128)) {
|
|
return handle_video_packet(session, &header, data + RTP_HEADER_SIZE, length - RTP_HEADER_SIZE, m->log);
|
|
}
|
|
|
|
// everything below here is for the old 16 bit protocol ------------------
|
|
|
|
if (header.data_length_lower == length - RTP_HEADER_SIZE) {
|
|
/* The message is sent in single part */
|
|
|
|
/* Message is not late; pick up the latest parameters */
|
|
session->rsequnum = header.sequnum;
|
|
session->rtimestamp = header.timestamp;
|
|
bwc_add_recv(session->bwc, length);
|
|
|
|
/* Invoke processing of active multiparted message */
|
|
if (session->mp != nullptr) {
|
|
session->mcb(session->m->mono_time, session->cs, session->mp);
|
|
session->mp = nullptr;
|
|
}
|
|
|
|
/* The message came in the allowed time;
|
|
*/
|
|
|
|
return session->mcb(session->m->mono_time, session->cs, new_message(&header, length - RTP_HEADER_SIZE,
|
|
data + RTP_HEADER_SIZE, length - RTP_HEADER_SIZE));
|
|
}
|
|
|
|
/* The message is sent in multiple parts */
|
|
|
|
if (session->mp != nullptr) {
|
|
/* There are 2 possible situations in this case:
|
|
* 1) being that we got the part of already processing message.
|
|
* 2) being that we got the part of a new/old message.
|
|
*
|
|
* We handle them differently as we only allow a single multiparted
|
|
* processing message
|
|
*/
|
|
if (session->mp->header.sequnum == header.sequnum &&
|
|
session->mp->header.timestamp == header.timestamp) {
|
|
/* First case */
|
|
|
|
/* Make sure we have enough allocated memory */
|
|
if (session->mp->header.data_length_lower - session->mp->len < length - RTP_HEADER_SIZE ||
|
|
session->mp->header.data_length_lower <= header.offset_lower) {
|
|
/* There happened to be some corruption on the stream;
|
|
* continue wihtout this part
|
|
*/
|
|
return 0;
|
|
}
|
|
|
|
memcpy(session->mp->data + header.offset_lower, data + RTP_HEADER_SIZE,
|
|
length - RTP_HEADER_SIZE);
|
|
session->mp->len += length - RTP_HEADER_SIZE;
|
|
bwc_add_recv(session->bwc, length);
|
|
|
|
if (session->mp->len == session->mp->header.data_length_lower) {
|
|
/* Received a full message; now push it for the further
|
|
* processing.
|
|
*/
|
|
session->mcb(session->m->mono_time, session->cs, session->mp);
|
|
session->mp = nullptr;
|
|
}
|
|
} else {
|
|
/* Second case */
|
|
if (session->mp->header.timestamp > header.timestamp) {
|
|
/* The received message part is from the old message;
|
|
* discard it.
|
|
*/
|
|
return 0;
|
|
}
|
|
|
|
/* Push the previous message for processing */
|
|
session->mcb(session->m->mono_time, session->cs, session->mp);
|
|
|
|
session->mp = nullptr;
|
|
goto NEW_MULTIPARTED;
|
|
}
|
|
} else {
|
|
/* In this case threat the message as if it was received in order
|
|
*/
|
|
/* This is also a point for new multiparted messages */
|
|
NEW_MULTIPARTED:
|
|
|
|
/* Message is not late; pick up the latest parameters */
|
|
session->rsequnum = header.sequnum;
|
|
session->rtimestamp = header.timestamp;
|
|
bwc_add_recv(session->bwc, length);
|
|
|
|
/* Store message.
|
|
*/
|
|
session->mp = new_message(&header, header.data_length_lower, data + RTP_HEADER_SIZE, length - RTP_HEADER_SIZE);
|
|
|
|
if (session->mp != nullptr) {
|
|
memmove(session->mp->data + header.offset_lower, session->mp->data, session->mp->len);
|
|
} else {
|
|
LOGGER_WARNING(m->log, "new_message() returned a null pointer");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
size_t rtp_header_pack(uint8_t *const rdata, const struct RTPHeader *header)
|
|
{
|
|
uint8_t *p = rdata;
|
|
*p = (header->ve & 3) << 6
|
|
| (header->pe & 1) << 5
|
|
| (header->xe & 1) << 4
|
|
| (header->cc & 0xf);
|
|
++p;
|
|
*p = (header->ma & 1) << 7
|
|
| (header->pt & 0x7f);
|
|
++p;
|
|
|
|
p += net_pack_u16(p, header->sequnum);
|
|
p += net_pack_u32(p, header->timestamp);
|
|
p += net_pack_u32(p, header->ssrc);
|
|
p += net_pack_u64(p, header->flags);
|
|
p += net_pack_u32(p, header->offset_full);
|
|
p += net_pack_u32(p, header->data_length_full);
|
|
p += net_pack_u32(p, header->received_length_full);
|
|
|
|
for (size_t i = 0; i < RTP_PADDING_FIELDS; ++i) {
|
|
p += net_pack_u32(p, 0);
|
|
}
|
|
|
|
p += net_pack_u16(p, header->offset_lower);
|
|
p += net_pack_u16(p, header->data_length_lower);
|
|
assert(p == rdata + RTP_HEADER_SIZE);
|
|
return p - rdata;
|
|
}
|
|
|
|
size_t rtp_header_unpack(const uint8_t *data, struct RTPHeader *header)
|
|
{
|
|
const uint8_t *p = data;
|
|
header->ve = (*p >> 6) & 3;
|
|
header->pe = (*p >> 5) & 1;
|
|
header->xe = (*p >> 4) & 1;
|
|
header->cc = *p & 0xf;
|
|
++p;
|
|
|
|
header->ma = (*p >> 7) & 1;
|
|
header->pt = *p & 0x7f;
|
|
++p;
|
|
|
|
p += net_unpack_u16(p, &header->sequnum);
|
|
p += net_unpack_u32(p, &header->timestamp);
|
|
p += net_unpack_u32(p, &header->ssrc);
|
|
p += net_unpack_u64(p, &header->flags);
|
|
p += net_unpack_u32(p, &header->offset_full);
|
|
p += net_unpack_u32(p, &header->data_length_full);
|
|
p += net_unpack_u32(p, &header->received_length_full);
|
|
|
|
p += sizeof(uint32_t) * RTP_PADDING_FIELDS;
|
|
|
|
p += net_unpack_u16(p, &header->offset_lower);
|
|
p += net_unpack_u16(p, &header->data_length_lower);
|
|
assert(p == data + RTP_HEADER_SIZE);
|
|
return p - data;
|
|
}
|
|
|
|
RTPSession *rtp_new(int payload_type, Messenger *m, Tox *tox, uint32_t friendnumber,
|
|
BWController *bwc, void *cs, rtp_m_cb *mcb)
|
|
{
|
|
assert(mcb != nullptr);
|
|
assert(cs != nullptr);
|
|
assert(m != nullptr);
|
|
|
|
RTPSession *session = (RTPSession *)calloc(1, sizeof(RTPSession));
|
|
|
|
if (session == nullptr) {
|
|
LOGGER_WARNING(m->log, "Alloc failed! Program might misbehave!");
|
|
return nullptr;
|
|
}
|
|
|
|
session->work_buffer_list = (struct RTPWorkBufferList *)calloc(1, sizeof(struct RTPWorkBufferList));
|
|
|
|
if (session->work_buffer_list == nullptr) {
|
|
LOGGER_ERROR(m->log, "out of memory while allocating work buffer list");
|
|
free(session);
|
|
return nullptr;
|
|
}
|
|
|
|
// First entry is free.
|
|
session->work_buffer_list->next_free_entry = 0;
|
|
|
|
session->ssrc = payload_type == RTP_TYPE_VIDEO ? 0 : random_u32(m->rng);
|
|
session->payload_type = payload_type;
|
|
session->m = m;
|
|
session->tox = tox;
|
|
session->friend_number = friendnumber;
|
|
|
|
// set NULL just in case
|
|
session->mp = nullptr;
|
|
session->first_packets_counter = 1;
|
|
|
|
/* Also set payload type as prefix */
|
|
session->bwc = bwc;
|
|
session->cs = cs;
|
|
session->mcb = mcb;
|
|
|
|
if (-1 == rtp_allow_receiving(session)) {
|
|
LOGGER_WARNING(m->log, "Failed to start rtp receiving mode");
|
|
free(session->work_buffer_list);
|
|
free(session);
|
|
return nullptr;
|
|
}
|
|
|
|
return session;
|
|
}
|
|
|
|
void rtp_kill(RTPSession *session)
|
|
{
|
|
if (session == nullptr) {
|
|
return;
|
|
}
|
|
|
|
LOGGER_DEBUG(session->m->log, "Terminated RTP session: %p", (void *)session);
|
|
rtp_stop_receiving(session);
|
|
|
|
LOGGER_DEBUG(session->m->log, "Terminated RTP session V3 work_buffer_list->next_free_entry: %d",
|
|
(int)session->work_buffer_list->next_free_entry);
|
|
|
|
for (int8_t i = 0; i < session->work_buffer_list->next_free_entry; ++i) {
|
|
free(session->work_buffer_list->work_buffer[i].buf);
|
|
}
|
|
free(session->work_buffer_list);
|
|
free(session);
|
|
}
|
|
|
|
int rtp_allow_receiving(RTPSession *session)
|
|
{
|
|
if (session == nullptr) {
|
|
return -1;
|
|
}
|
|
|
|
if (m_callback_rtp_packet(session->m, session->friend_number, session->payload_type,
|
|
handle_rtp_packet, session) == -1) {
|
|
LOGGER_WARNING(session->m->log, "Failed to register rtp receive handler");
|
|
return -1;
|
|
}
|
|
|
|
LOGGER_DEBUG(session->m->log, "Started receiving on session: %p", (void *)session);
|
|
return 0;
|
|
}
|
|
|
|
int rtp_stop_receiving(RTPSession *session)
|
|
{
|
|
if (session == nullptr) {
|
|
return -1;
|
|
}
|
|
|
|
m_callback_rtp_packet(session->m, session->friend_number, session->payload_type, nullptr, nullptr);
|
|
|
|
LOGGER_DEBUG(session->m->log, "Stopped receiving on session: %p", (void *)session);
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Send a frame of audio or video data, chunked in @ref RTPMessage instances.
|
|
*
|
|
* @param session The A/V session to send the data for.
|
|
* @param data A byte array of length @p length.
|
|
* @param length The number of bytes to send from @p data.
|
|
* @param is_keyframe Whether this video frame is a key frame. If it is an
|
|
* audio frame, this parameter is ignored.
|
|
*/
|
|
int rtp_send_data(RTPSession *session, const uint8_t *data, uint32_t length,
|
|
bool is_keyframe, const Logger *log)
|
|
{
|
|
if (session == nullptr) {
|
|
LOGGER_ERROR(log, "No session!");
|
|
return -1;
|
|
}
|
|
|
|
struct RTPHeader header = {0};
|
|
|
|
header.ve = 2; // this is unused in toxav
|
|
|
|
header.pe = 0;
|
|
|
|
header.xe = 0;
|
|
|
|
header.cc = 0;
|
|
|
|
header.ma = 0;
|
|
|
|
header.pt = session->payload_type % 128;
|
|
|
|
header.sequnum = session->sequnum;
|
|
|
|
header.timestamp = current_time_monotonic(session->m->mono_time);
|
|
|
|
header.ssrc = session->ssrc;
|
|
|
|
header.offset_lower = 0;
|
|
|
|
// here the highest bits gets stripped anyway, no need to do keyframe bit magic here!
|
|
header.data_length_lower = length;
|
|
|
|
if (session->payload_type == RTP_TYPE_VIDEO) {
|
|
header.flags = RTP_LARGE_FRAME;
|
|
}
|
|
|
|
uint16_t length_safe = (uint16_t)length;
|
|
|
|
if (length > UINT16_MAX) {
|
|
length_safe = UINT16_MAX;
|
|
}
|
|
|
|
header.data_length_lower = length_safe;
|
|
header.data_length_full = length; // without header
|
|
header.offset_lower = 0;
|
|
header.offset_full = 0;
|
|
|
|
if (is_keyframe) {
|
|
header.flags |= RTP_KEY_FRAME;
|
|
}
|
|
|
|
VLA(uint8_t, rdata, length + RTP_HEADER_SIZE + 1);
|
|
memset(rdata, 0, SIZEOF_VLA(rdata));
|
|
rdata[0] = session->payload_type; // packet id == payload_type
|
|
|
|
if (MAX_CRYPTO_DATA_SIZE > (length + RTP_HEADER_SIZE + 1)) {
|
|
/*
|
|
* The length is lesser than the maximum allowed length (including header)
|
|
* Send the packet in single piece.
|
|
*/
|
|
rtp_header_pack(rdata + 1, &header);
|
|
memcpy(rdata + 1 + RTP_HEADER_SIZE, data, length);
|
|
|
|
if (-1 == rtp_send_custom_lossy_packet(session->tox, session->friend_number, rdata, SIZEOF_VLA(rdata))) {
|
|
char *netstrerror = net_new_strerror(net_error());
|
|
LOGGER_WARNING(session->m->log, "RTP send failed (len: %u)! net error: %s",
|
|
(unsigned)SIZEOF_VLA(rdata), netstrerror);
|
|
net_kill_strerror(netstrerror);
|
|
}
|
|
} else {
|
|
/*
|
|
* The length is greater than the maximum allowed length (including header)
|
|
* Send the packet in multiple pieces.
|
|
*/
|
|
uint32_t sent = 0;
|
|
uint16_t piece = MAX_CRYPTO_DATA_SIZE - (RTP_HEADER_SIZE + 1);
|
|
|
|
while ((length - sent) + RTP_HEADER_SIZE + 1 > MAX_CRYPTO_DATA_SIZE) {
|
|
rtp_header_pack(rdata + 1, &header);
|
|
memcpy(rdata + 1 + RTP_HEADER_SIZE, data + sent, piece);
|
|
|
|
if (-1 == rtp_send_custom_lossy_packet(session->tox, session->friend_number,
|
|
rdata, piece + RTP_HEADER_SIZE + 1)) {
|
|
char *netstrerror = net_new_strerror(net_error());
|
|
LOGGER_WARNING(session->m->log, "RTP send failed (len: %d)! net error: %s",
|
|
piece + RTP_HEADER_SIZE + 1, netstrerror);
|
|
net_kill_strerror(netstrerror);
|
|
}
|
|
|
|
sent += piece;
|
|
header.offset_lower = sent;
|
|
header.offset_full = sent; // raw data offset, without any header
|
|
}
|
|
|
|
/* Send remaining */
|
|
piece = length - sent;
|
|
|
|
if (piece != 0) {
|
|
rtp_header_pack(rdata + 1, &header);
|
|
memcpy(rdata + 1 + RTP_HEADER_SIZE, data + sent, piece);
|
|
|
|
if (-1 == rtp_send_custom_lossy_packet(session->tox, session->friend_number, rdata,
|
|
piece + RTP_HEADER_SIZE + 1)) {
|
|
char *netstrerror = net_new_strerror(net_error());
|
|
LOGGER_WARNING(session->m->log, "RTP send failed (len: %d)! net error: %s",
|
|
piece + RTP_HEADER_SIZE + 1, netstrerror);
|
|
net_kill_strerror(netstrerror);
|
|
}
|
|
}
|
|
}
|
|
|
|
++session->sequnum;
|
|
return 0;
|
|
}
|