Index: webrtc/call/rtp_rtcp_demuxer_helper.cc |
diff --git a/webrtc/call/rtp_rtcp_demuxer_helper.cc b/webrtc/call/rtp_rtcp_demuxer_helper.cc |
deleted file mode 100644 |
index e8d3cbfadbe94b46b5cc6ae7b4009b1401588e5d..0000000000000000000000000000000000000000 |
--- a/webrtc/call/rtp_rtcp_demuxer_helper.cc |
+++ /dev/null |
@@ -1,55 +0,0 @@ |
-/* |
- * Copyright (c) 2017 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/call/rtp_rtcp_demuxer_helper.h" |
- |
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
- |
-namespace webrtc { |
- |
-rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc( |
- rtc::ArrayView<const uint8_t> packet) { |
- rtcp::CommonHeader header; |
- for (const uint8_t* next_packet = packet.begin(); next_packet < packet.end(); |
- next_packet = header.NextPacket()) { |
- if (!header.Parse(next_packet, packet.end() - next_packet)) { |
- return rtc::Optional<uint32_t>(); |
- } |
- |
- switch (header.type()) { |
- case rtcp::Bye::kPacketType: |
- case rtcp::ExtendedReports::kPacketType: |
- case rtcp::Psfb::kPacketType: |
- case rtcp::ReceiverReport::kPacketType: |
- case rtcp::Rtpfb::kPacketType: |
- case rtcp::SenderReport::kPacketType: { |
- // Sender SSRC at the beginning of the RTCP payload. |
- if (header.payload_size_bytes() >= sizeof(uint32_t)) { |
- const uint32_t ssrc_sender = |
- ByteReader<uint32_t>::ReadBigEndian(header.payload()); |
- return rtc::Optional<uint32_t>(ssrc_sender); |
- } else { |
- return rtc::Optional<uint32_t>(); |
- } |
- } |
- } |
- } |
- |
- return rtc::Optional<uint32_t>(); |
-} |
- |
-} // namespace webrtc |