2bb1bdab8d
A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
120 lines
2.9 KiB
C++
120 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/vie_sender.h"
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#include <assert.h>
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/utility/interface/rtp_dump.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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namespace webrtc {
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ViESender::ViESender(int channel_id)
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: channel_id_(channel_id),
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critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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transport_(NULL),
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rtp_dump_(NULL) {
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}
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ViESender::~ViESender() {
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if (rtp_dump_) {
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rtp_dump_->Stop();
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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = NULL;
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}
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}
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int ViESender::RegisterSendTransport(Transport* transport) {
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CriticalSectionScoped cs(critsect_.get());
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if (transport_) {
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return -1;
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}
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transport_ = transport;
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return 0;
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}
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int ViESender::DeregisterSendTransport() {
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CriticalSectionScoped cs(critsect_.get());
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if (transport_ == NULL) {
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return -1;
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}
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transport_ = NULL;
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return 0;
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}
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int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
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CriticalSectionScoped cs(critsect_.get());
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if (rtp_dump_) {
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// Packet dump is already started, restart it.
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rtp_dump_->Stop();
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} else {
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rtp_dump_ = RtpDump::CreateRtpDump();
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if (rtp_dump_ == NULL) {
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return -1;
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}
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}
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if (rtp_dump_->Start(file_nameUTF8) != 0) {
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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = NULL;
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return -1;
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}
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return 0;
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}
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int ViESender::StopRTPDump() {
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CriticalSectionScoped cs(critsect_.get());
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if (rtp_dump_) {
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if (rtp_dump_->IsActive()) {
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rtp_dump_->Stop();
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}
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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = NULL;
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} else {
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return -1;
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}
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return 0;
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}
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int ViESender::SendPacket(int vie_id, const void* data, int len) {
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CriticalSectionScoped cs(critsect_.get());
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if (!transport_) {
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// No transport
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return -1;
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}
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assert(ChannelId(vie_id) == channel_id_);
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if (rtp_dump_) {
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rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
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static_cast<uint16_t>(len));
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}
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return transport_->SendPacket(channel_id_, data, len);
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}
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int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
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CriticalSectionScoped cs(critsect_.get());
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if (!transport_) {
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return -1;
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}
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assert(ChannelId(vie_id) == channel_id_);
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if (rtp_dump_) {
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rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
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static_cast<uint16_t>(len));
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}
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return transport_->SendRTCPPacket(channel_id_, data, len);
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}
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} // namespace webrtc
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