Repo created

TMessagesProj/jni/voip/webrtc/call/rtx_receive_stream.cc

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+/*
+ *  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 "call/rtx_receive_stream.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "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) {
+  packet_checker_.Detach();
+  if (associated_payload_types_.empty()) {
+    RTC_LOG(LS_WARNING)
+        << "RtxReceiveStream created with empty payload type mapping.";
+  }
+}
+
+RtxReceiveStream::~RtxReceiveStream() = default;
+
+void RtxReceiveStream::SetAssociatedPayloadTypes(
+    std::map<int, int> associated_payload_types) {
+  RTC_DCHECK_RUN_ON(&packet_checker_);
+  associated_payload_types_ = std::move(associated_payload_types);
+}
+
+void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
+  RTC_DCHECK_RUN_ON(&packet_checker_);
+  if (rtp_receive_statistics_) {
+    rtp_receive_statistics_->OnRtpPacket(rtx_packet);
+  }
+  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()) {
+    RTC_DLOG(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);
+  media_packet.set_arrival_time(rtx_packet.arrival_time());
+
+  // 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