Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/call/rtx_receive_stream.cc
+++ b/call/rtx_receive_stream.cc
@ -0,0 +1,77 @@
+/*
+ *  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 <utility>
+
+#include "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/rtc_base/logging.h"
+
+namespace webrtc {
+
+RtxReceiveStream::RtxReceiveStream(
+    RtpPacketSinkInterface* media_sink,
+    std::map<int, int> associated_payload_types,
+    uint32_t media_ssrc,
+    ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
+    : media_sink_(media_sink),
+      associated_payload_types_(std::move(associated_payload_types)),
+      media_ssrc_(media_ssrc),
+      rtp_receive_statistics_(rtp_receive_statistics) {
+  if (associated_payload_types_.empty()) {
+    LOG(LS_WARNING)
+        << "RtxReceiveStream created with empty payload type mapping.";
+  }
+}
+
+RtxReceiveStream::~RtxReceiveStream() = default;
+
+void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
+  if (rtp_receive_statistics_) {
+    RTPHeader header;
+    rtx_packet.GetHeader(&header);
+    rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
+                                            false /* retransmitted */);
+  }
+  rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
+
+  if (payload.size() < kRtxHeaderSize) {
+    return;
+  }
+
+  auto it = associated_payload_types_.find(rtx_packet.PayloadType());
+  if (it == associated_payload_types_.end()) {
+    LOG(LS_VERBOSE) << "Unknown payload type "
+                    << static_cast<int>(rtx_packet.PayloadType())
+                    << " on rtx ssrc " << rtx_packet.Ssrc();
+    return;
+  }
+  RtpPacketReceived media_packet;
+  media_packet.CopyHeaderFrom(rtx_packet);
+
+  media_packet.SetSsrc(media_ssrc_);
+  media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
+  media_packet.SetPayloadType(it->second);
+  media_packet.set_recovered(true);
+
+  // Skip the RTX header.
+  rtc::ArrayView<const uint8_t> rtx_payload =
+      payload.subview(kRtxHeaderSize);
+
+  uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
+  RTC_DCHECK(media_payload != nullptr);
+
+  memcpy(media_payload, rtx_payload.data(), rtx_payload.size());
+
+  media_sink_->OnRtpPacket(media_packet);
+}
+
+}  // namespace webrtc