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      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 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
     15 
     16 #include "webrtc/modules/utility/interface/rtp_dump.h"
     17 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
     18 #include "webrtc/system_wrappers/interface/trace.h"
     19 
     20 namespace webrtc {
     21 
     22 ViESender::ViESender(int channel_id)
     23     : channel_id_(channel_id),
     24       critsect_(CriticalSectionWrapper::CreateCriticalSection()),
     25       transport_(NULL),
     26       rtp_dump_(NULL) {
     27 }
     28 
     29 ViESender::~ViESender() {
     30   if (rtp_dump_) {
     31     rtp_dump_->Stop();
     32     RtpDump::DestroyRtpDump(rtp_dump_);
     33     rtp_dump_ = NULL;
     34   }
     35 }
     36 
     37 int ViESender::RegisterSendTransport(Transport* transport) {
     38   CriticalSectionScoped cs(critsect_.get());
     39   if (transport_) {
     40     return -1;
     41   }
     42   transport_ = transport;
     43   return 0;
     44 }
     45 
     46 int ViESender::DeregisterSendTransport() {
     47   CriticalSectionScoped cs(critsect_.get());
     48   if (transport_ == NULL) {
     49     return -1;
     50   }
     51   transport_ = NULL;
     52   return 0;
     53 }
     54 
     55 int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
     56   CriticalSectionScoped cs(critsect_.get());
     57   if (rtp_dump_) {
     58     // Packet dump is already started, restart it.
     59     rtp_dump_->Stop();
     60   } else {
     61     rtp_dump_ = RtpDump::CreateRtpDump();
     62     if (rtp_dump_ == NULL) {
     63       return -1;
     64     }
     65   }
     66   if (rtp_dump_->Start(file_nameUTF8) != 0) {
     67     RtpDump::DestroyRtpDump(rtp_dump_);
     68     rtp_dump_ = NULL;
     69     return -1;
     70   }
     71   return 0;
     72 }
     73 
     74 int ViESender::StopRTPDump() {
     75   CriticalSectionScoped cs(critsect_.get());
     76   if (rtp_dump_) {
     77     if (rtp_dump_->IsActive()) {
     78       rtp_dump_->Stop();
     79     }
     80     RtpDump::DestroyRtpDump(rtp_dump_);
     81     rtp_dump_ = NULL;
     82   } else {
     83     return -1;
     84   }
     85   return 0;
     86 }
     87 
     88 int ViESender::SendPacket(int vie_id, const void* data, int len) {
     89   CriticalSectionScoped cs(critsect_.get());
     90   if (!transport_) {
     91     // No transport
     92     return -1;
     93   }
     94   assert(ChannelId(vie_id) == channel_id_);
     95 
     96   if (rtp_dump_) {
     97     rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
     98                           static_cast<uint16_t>(len));
     99   }
    100 
    101   return transport_->SendPacket(channel_id_, data, len);
    102 }
    103 
    104 int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
    105   CriticalSectionScoped cs(critsect_.get());
    106   if (!transport_) {
    107     return -1;
    108   }
    109   assert(ChannelId(vie_id) == channel_id_);
    110 
    111   if (rtp_dump_) {
    112     rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
    113                           static_cast<uint16_t>(len));
    114   }
    115 
    116   return transport_->SendRTCPPacket(channel_id_, data, len);
    117 }
    118 
    119 }  // namespace webrtc
    120