Home | History | Annotate | Download | only in video_engine 
      1 /*
      2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #include "webrtc/video_engine/vie_sender.h"
     12 
     13 #include <assert.h>
     14 
     15 #include "webrtc/modules/utility/interface/rtp_dump.h"
     16 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
     17 #include "webrtc/system_wrappers/interface/trace.h"
     18 
     19 namespace webrtc {
     20 
     21 ViESender::ViESender(int channel_id)
     22     : channel_id_(channel_id),
     23       critsect_(CriticalSectionWrapper::CreateCriticalSection()),
     24       transport_(NULL),
     25       rtp_dump_(NULL) {
     26 }
     27 
     28 ViESender::~ViESender() {
     29   if (rtp_dump_) {
     30     rtp_dump_->Stop();
     31     RtpDump::DestroyRtpDump(rtp_dump_);
     32     rtp_dump_ = NULL;
     33   }
     34 }
     35 
     36 int ViESender::RegisterSendTransport(Transport* transport) {
     37   CriticalSectionScoped cs(critsect_.get());
     38   if (transport_) {
     39     return -1;
     40   }
     41   transport_ = transport;
     42   return 0;
     43 }
     44 
     45 int ViESender::DeregisterSendTransport() {
     46   CriticalSectionScoped cs(critsect_.get());
     47   if (transport_ == NULL) {
     48     return -1;
     49   }
     50   transport_ = NULL;
     51   return 0;
     52 }
     53 
     54 int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
     55   CriticalSectionScoped cs(critsect_.get());
     56   if (rtp_dump_) {
     57     // Packet dump is already started, restart it.
     58     rtp_dump_->Stop();
     59   } else {
     60     rtp_dump_ = RtpDump::CreateRtpDump();
     61     if (rtp_dump_ == NULL) {
     62       return -1;
     63     }
     64   }
     65   if (rtp_dump_->Start(file_nameUTF8) != 0) {
     66     RtpDump::DestroyRtpDump(rtp_dump_);
     67     rtp_dump_ = NULL;
     68     return -1;
     69   }
     70   return 0;
     71 }
     72 
     73 int ViESender::StopRTPDump() {
     74   CriticalSectionScoped cs(critsect_.get());
     75   if (rtp_dump_) {
     76     if (rtp_dump_->IsActive()) {
     77       rtp_dump_->Stop();
     78     }
     79     RtpDump::DestroyRtpDump(rtp_dump_);
     80     rtp_dump_ = NULL;
     81   } else {
     82     return -1;
     83   }
     84   return 0;
     85 }
     86 
     87 int ViESender::SendPacket(int vie_id, const void* data, int len) {
     88   CriticalSectionScoped cs(critsect_.get());
     89   if (!transport_) {
     90     // No transport
     91     return -1;
     92   }
     93   assert(ChannelId(vie_id) == channel_id_);
     94 
     95   if (rtp_dump_) {
     96     rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
     97                           static_cast<uint16_t>(len));
     98   }
     99 
    100   return transport_->SendPacket(channel_id_, data, len);
    101 }
    102 
    103 int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
    104   CriticalSectionScoped cs(critsect_.get());
    105   if (!transport_) {
    106     return -1;
    107   }
    108   assert(ChannelId(vie_id) == channel_id_);
    109 
    110   if (rtp_dump_) {
    111     rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
    112                           static_cast<uint16_t>(len));
    113   }
    114 
    115   return transport_->SendRTCPPacket(channel_id_, data, len);
    116 }
    117 
    118 }  // namespace webrtc
    119