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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/video_engine/vie_sender.h"
#include <cassert>
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
ViESender::ViESender(int channel_id)
: channel_id_(channel_id),
critsect_(CriticalSectionWrapper::CreateCriticalSection()),
external_encryption_(NULL),
encryption_buffer_(NULL),
transport_(NULL),
rtp_dump_(NULL) {
}
ViESender::~ViESender() {
if (encryption_buffer_) {
delete[] encryption_buffer_;
encryption_buffer_ = NULL;
}
if (rtp_dump_) {
rtp_dump_->Stop();
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
}
}
int ViESender::RegisterExternalEncryption(Encryption* encryption) {
CriticalSectionScoped cs(critsect_.get());
if (external_encryption_) {
return -1;
}
encryption_buffer_ = new uint8_t[kViEMaxMtu];
if (encryption_buffer_ == NULL) {
return -1;
}
external_encryption_ = encryption;
return 0;
}
int ViESender::DeregisterExternalEncryption() {
CriticalSectionScoped cs(critsect_.get());
if (external_encryption_ == NULL) {
return -1;
}
if (encryption_buffer_) {
delete[] encryption_buffer_;
encryption_buffer_ = NULL;
}
external_encryption_ = NULL;
return 0;
}
int ViESender::RegisterSendTransport(Transport* transport) {
CriticalSectionScoped cs(critsect_.get());
if (transport_) {
return -1;
}
transport_ = transport;
return 0;
}
int ViESender::DeregisterSendTransport() {
CriticalSectionScoped cs(critsect_.get());
if (transport_ == NULL) {
return -1;
}
transport_ = NULL;
return 0;
}
int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
CriticalSectionScoped cs(critsect_.get());
if (rtp_dump_) {
// Packet dump is already started, restart it.
rtp_dump_->Stop();
} else {
rtp_dump_ = RtpDump::CreateRtpDump();
if (rtp_dump_ == NULL) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
"StartSRTPDump: Failed to create RTP dump");
return -1;
}
}
if (rtp_dump_->Start(file_nameUTF8) != 0) {
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
"StartRTPDump: Failed to start RTP dump");
return -1;
}
return 0;
}
int ViESender::StopRTPDump() {
CriticalSectionScoped cs(critsect_.get());
if (rtp_dump_) {
if (rtp_dump_->IsActive()) {
rtp_dump_->Stop();
} else {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
"StopRTPDump: Dump not active");
}
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
} else {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
"StopRTPDump: RTP dump not started");
return -1;
}
return 0;
}
int ViESender::SendPacket(int vie_id, const void* data, int len) {
CriticalSectionScoped cs(critsect_.get());
if (!transport_) {
// No transport
return -1;
}
assert(ChannelId(vie_id) == channel_id_);
// TODO(mflodman) Change decrypt to get rid of this cast.
void* tmp_ptr = const_cast<void*>(data);
unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
// Data length for packets sent to possible encryption and to the transport.
int send_packet_length = len;
if (rtp_dump_) {
rtp_dump_->DumpPacket(send_packet, send_packet_length);
}
if (external_encryption_) {
// Encryption buffer size.
int encrypted_packet_length = kViEMaxMtu;
external_encryption_->encrypt(channel_id_, send_packet, encryption_buffer_,
send_packet_length, &encrypted_packet_length);
send_packet = encryption_buffer_;
send_packet_length = encrypted_packet_length;
}
const int bytes_sent = transport_->SendPacket(channel_id_, send_packet,
send_packet_length);
if (bytes_sent != send_packet_length) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, channel_id_,
"ViESender::SendPacket - Transport failed to send RTP packet");
}
return bytes_sent;
}
int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
CriticalSectionScoped cs(critsect_.get());
if (!transport_) {
return -1;
}
assert(ChannelId(vie_id) == channel_id_);
// Prepare for possible encryption and sending.
// TODO(mflodman) Change decrypt to get rid of this cast.
void* tmp_ptr = const_cast<void*>(data);
unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
// Data length for packets sent to possible encryption and to the transport.
int send_packet_length = len;
if (rtp_dump_) {
rtp_dump_->DumpPacket(send_packet, send_packet_length);
}
if (external_encryption_) {
// Encryption buffer size.
int encrypted_packet_length = kViEMaxMtu;
external_encryption_->encrypt_rtcp(
channel_id_, send_packet, encryption_buffer_, send_packet_length,
&encrypted_packet_length);
send_packet = encryption_buffer_;
send_packet_length = encrypted_packet_length;
}
const int bytes_sent = transport_->SendRTCPPacket(channel_id_, send_packet,
send_packet_length);
if (bytes_sent != send_packet_length) {
WEBRTC_TRACE(
webrtc::kTraceWarning, webrtc::kTraceVideo, channel_id_,
"ViESender::SendRTCPPacket - Transport failed to send RTCP packet"
" (%d vs %d)", bytes_sent, send_packet_length);
}
return bytes_sent;
}
} // namespace webrtc
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