Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2014 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/modules/rtp_rtcp/source/rtp_format.h"
      12             : 
      13             : #include <utility>
      14             : 
      15             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
      16             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
      17             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
      18             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
      19             : 
      20             : namespace webrtc {
      21           0 : RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
      22             :                                      size_t max_payload_len,
      23             :                                      const RTPVideoTypeHeader* rtp_type_header,
      24             :                                      FrameType frame_type) {
      25           0 :   switch (type) {
      26             :     case kRtpVideoH264:
      27           0 :       RTC_CHECK(rtp_type_header);
      28             :       return new RtpPacketizerH264(max_payload_len,
      29           0 :                                    rtp_type_header->H264.packetization_mode);
      30             :     case kRtpVideoVp8:
      31           0 :       RTC_CHECK(rtp_type_header);
      32           0 :       return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
      33             :     case kRtpVideoVp9:
      34           0 :       RTC_CHECK(rtp_type_header);
      35           0 :       return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len);
      36             :     case kRtpVideoGeneric:
      37           0 :       return new RtpPacketizerGeneric(frame_type, max_payload_len);
      38             :     case kRtpVideoNone:
      39           0 :       RTC_NOTREACHED();
      40             :   }
      41           0 :   return NULL;
      42             : }
      43             : 
      44           0 : RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
      45           0 :   switch (type) {
      46             :     case kRtpVideoH264:
      47           0 :       return new RtpDepacketizerH264();
      48             :     case kRtpVideoVp8:
      49           0 :       return new RtpDepacketizerVp8();
      50             :     case kRtpVideoVp9:
      51           0 :       return new RtpDepacketizerVp9();
      52             :     case kRtpVideoGeneric:
      53           0 :       return new RtpDepacketizerGeneric();
      54             :     case kRtpVideoNone:
      55           0 :       assert(false);
      56             :   }
      57           0 :   return NULL;
      58             : }
      59             : }  // namespace webrtc