Integration of VP9 packetization.

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

 /*
  *  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/modules/rtp_rtcp/source/rtp_sender_video.h"
 
 #include <stdlib.h>
 #include <string.h>
 
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };
 
 struct RtpPacket {
   uint16_t rtpHeaderLength;
   ForwardErrorCorrection::Packet* pkt;
@@ skipping 35 matching lines @@
 }
 
 // Static.
 RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
     const uint32_t maxBitRate) {
   RtpVideoCodecTypes videoType = kRtpVideoGeneric;
   if (RtpUtility::StringCompare(payloadName, "VP8", 3)) {
     videoType = kRtpVideoVp8;
+  } else if (RtpUtility::StringCompare(payloadName, "VP9", 3)) {
+    videoType = kRtpVideoVp9;
   } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) {
     videoType = kRtpVideoH264;
   } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) {
     videoType = kRtpVideoGeneric;
   } else {
     videoType = kRtpVideoGeneric;
   }
   RtpUtility::Payload* payload = new RtpUtility::Payload();
   payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
   strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
@@ skipping 278 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   CriticalSectionScoped cs(crit_.get());
   _retransmissionSettings = settings;
 }
 
 }  // namespace webrtc