From 075c20fe1626761c5fe545434f1e0cdc9141fea5 Mon Sep 17 00:00:00 2001
From: Per K <perkj@webrtc.org>
Date: Thu, 5 Jan 2023 15:20:33 +0100
Subject: [PATCH] Implement FakeCall::DeliverRtpPacket and DeliverRtcpPacket

This is done in peparation for using these methods in the engines.

Bug: webrtc:7135, webrtc:14795
Change-Id: I1255c035437d31398327318c3dbd73e70a11a5cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290577
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39044}
---
 media/BUILD.gn                   |  1 +
 media/engine/fake_webrtc_call.cc | 52 ++++++++++++++++++++++++--------
 media/engine/fake_webrtc_call.h  | 12 ++++++++
 3 files changed, 53 insertions(+), 12 deletions(-)

diff --git a/media/BUILD.gn b/media/BUILD.gn
index 8ed603127a..4ba6f382bf 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -707,6 +707,7 @@ if (rtc_include_tests) {
       "../api/task_queue",
       "../api/task_queue:pending_task_safety_flag",
       "../api/transport:field_trial_based_config",
+      "../api/units:timestamp",
       "../api/video:encoded_image",
       "../api/video:video_bitrate_allocation",
       "../api/video:video_frame",
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index 8046c3ad3a..c2a068d2f2 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -10,11 +10,14 @@
 
 #include "media/engine/fake_webrtc_call.h"
 
+#include <cstdint>
 #include <utility>
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
 #include "api/call/audio_sink.h"
+#include "api/units/timestamp.h"
+#include "call/packet_receiver.h"
 #include "media/base/media_channel.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/checks.h"
@@ -667,36 +670,61 @@ webrtc::PacketReceiver* FakeCall::Receiver() {
   return this;
 }
 
-FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,
-                                                 rtc::CopyOnWriteBuffer packet,
-                                                 int64_t packet_time_us) {
+webrtc::PacketReceiver::DeliveryStatus FakeCall::DeliverPacket(
+    webrtc::MediaType media_type,
+    rtc::CopyOnWriteBuffer packet,
+    int64_t packet_time_us) {
+  RTC_DCHECK(webrtc::IsRtpPacket(packet));
+  uint32_t ssrc = ParseRtpSsrc(packet);
+  webrtc::Timestamp arrival_time =
+      packet_time_us > -1 ? webrtc::Timestamp::Micros(packet_time_us)
+                          : webrtc::Timestamp::Zero();
+  if (DeliverPacketInternal(media_type, ssrc, packet, arrival_time)) {
+    return DELIVERY_OK;
+  }
+  return DELIVERY_UNKNOWN_SSRC;
+}
+
+void FakeCall::DeliverRtpPacket(
+    webrtc::MediaType media_type,
+    webrtc::RtpPacketReceived packet,
+    OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  if (!DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
+                             packet.arrival_time())) {
+    if (undemuxable_packet_handler(packet)) {
+      DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
+                            packet.arrival_time());
+    }
+  }
+}
+
+bool FakeCall::DeliverPacketInternal(webrtc::MediaType media_type,
+                                     uint32_t ssrc,
+                                     const rtc::CopyOnWriteBuffer& packet,
+                                     webrtc::Timestamp arrival_time) {
   EXPECT_GE(packet.size(), 12u);
+  RTC_DCHECK(arrival_time.IsFinite());
   RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
              media_type == webrtc::MediaType::VIDEO);
 
-  if (!webrtc::IsRtpPacket(packet)) {
-    return DELIVERY_PACKET_ERROR;
-  }
-
-  uint32_t ssrc = ParseRtpSsrc(packet);
   if (media_type == webrtc::MediaType::VIDEO) {
     for (auto receiver : video_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
         ++delivered_packets_by_ssrc_[ssrc];
-        return DELIVERY_OK;
+        return true;
       }
     }
   }
   if (media_type == webrtc::MediaType::AUDIO) {
     for (auto receiver : audio_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
-        receiver->DeliverRtp(packet.cdata(), packet.size(), packet_time_us);
+        receiver->DeliverRtp(packet.cdata(), packet.size(), arrival_time.us());
         ++delivered_packets_by_ssrc_[ssrc];
-        return DELIVERY_OK;
+        return true;
       }
     }
   }
-  return DELIVERY_UNKNOWN_SSRC;
+  return false;
 }
 
 void FakeCall::SetStats(const webrtc::Call::Stats& stats) {
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 07b33c3a01..5fc2745316 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -444,6 +444,18 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
                                rtc::CopyOnWriteBuffer packet,
                                int64_t packet_time_us) override;
 
+  void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override {}
+
+  void DeliverRtpPacket(
+      webrtc::MediaType media_type,
+      webrtc::RtpPacketReceived packet,
+      OnUndemuxablePacketHandler un_demuxable_packet_handler) override;
+
+  bool DeliverPacketInternal(webrtc::MediaType media_type,
+                             uint32_t ssrc,
+                             const rtc::CopyOnWriteBuffer& packet,
+                             webrtc::Timestamp arrival_time);
+
   webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
       override {
     return &transport_controller_send_;