Delete legacy RtpHeaderParser wrapper

Bug: None
Change-Id: I4deec4fab631488ef2d0706848cbbe4e085825bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221617
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34341}

8 files changed, 36 insertions, 160 deletions

diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 08f2742528..b841228a8e 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -243,6 +243,7 @@ if (!is_component_build) {
       "../call:call_interfaces",
       "../common_video",
       "../media:rtc_internal_video_codecs",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
       "../modules/video_coding:video_coding_utility",
       "../rtc_base:checks",
       "../rtc_base:rtc_json",
diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc
index 08f4c7accc..d96c7be410 100644
--- a/rtc_tools/video_replay.cc
+++ b/rtc_tools/video_replay.cc
@@ -25,6 +25,7 @@
 #include "call/call.h"
 #include "common_video/libyuv/include/webrtc_libyuv.h"
 #include "media/engine/internal_decoder_factory.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/utility/ivf_file_writer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/string_to_number.h"
@@ -39,7 +40,6 @@
 #include "test/gtest.h"
 #include "test/null_transport.h"
 #include "test/rtp_file_reader.h"
-#include "test/rtp_header_parser.h"
 #include "test/run_loop.h"
 #include "test/run_test.h"
 #include "test/test_video_capturer.h"
@@ -544,11 +544,11 @@ class RtpReplayer final {
       if (!rtp_reader->NextPacket(&packet)) {
         break;
       }
-      RTPHeader header;
-      std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::CreateForTest());
-      parser->Parse(packet.data, packet.length, &header);
-      if (header.timestamp < start_timestamp ||
-          header.timestamp > stop_timestamp) {
+      rtc::CopyOnWriteBuffer packet_buffer(packet.data, packet.length);
+      RtpPacket header;
+      header.Parse(packet_buffer);
+      if (header.Timestamp() < start_timestamp ||
+          header.Timestamp() > stop_timestamp) {
         continue;
       }
 
@@ -560,10 +560,9 @@ class RtpReplayer final {
       ++num_packets;
       PacketReceiver::DeliveryStatus result = PacketReceiver::DELIVERY_OK;
       worker_thread->PostTask(ToQueuedTask([&]() {
-        result = call->Receiver()->DeliverPacket(
-            webrtc::MediaType::VIDEO,
-            rtc::CopyOnWriteBuffer(packet.data, packet.length),
-            /* packet_time_us */ -1);
+        result = call->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
+                                                 std::move(packet_buffer),
+                                                 /* packet_time_us */ -1);
         event.Set();
       }));
       event.Wait(/*give_up_after_ms=*/10000);
@@ -571,21 +570,17 @@ class RtpReplayer final {
         case PacketReceiver::DELIVERY_OK:
           break;
         case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
-          if (unknown_packets[header.ssrc] == 0)
-            fprintf(stderr, "Unknown SSRC: %u!\n", header.ssrc);
-          ++unknown_packets[header.ssrc];
+          if (unknown_packets[header.Ssrc()] == 0)
+            fprintf(stderr, "Unknown SSRC: %u!\n", header.Ssrc());
+          ++unknown_packets[header.Ssrc()];
           break;
         }
         case PacketReceiver::DELIVERY_PACKET_ERROR: {
           fprintf(stderr,
                   "Packet error, corrupt packets or incorrect setup?\n");
-          RTPHeader header;
-          std::unique_ptr<RtpHeaderParser> parser(
-              RtpHeaderParser::CreateForTest());
-          parser->Parse(packet.data, packet.length, &header);
           fprintf(stderr, "Packet len=%zu pt=%u seq=%u ts=%u ssrc=0x%8x\n",
-                  packet.length, header.payloadType, header.sequenceNumber,
-                  header.timestamp, header.ssrc);
+                  packet.length, header.PayloadType(), header.SequenceNumber(),
+                  header.Timestamp(), header.Ssrc());
           break;
         }
       }
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 988d15fd30..77f003cfbe 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -212,6 +212,7 @@ rtc_library("rtp_test_utils") {
     "../rtc_base/synchronization:mutex",
     "../rtc_base/system:arch",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("field_trial") {
diff --git a/test/fuzzers/rtp_header_parser_fuzzer.cc b/test/fuzzers/rtp_header_parser_fuzzer.cc
index d6af5ca3ce..cb5bea2456 100644
--- a/test/fuzzers/rtp_header_parser_fuzzer.cc
+++ b/test/fuzzers/rtp_header_parser_fuzzer.cc
@@ -22,27 +22,6 @@ namespace webrtc {
 void FuzzOneInput(const uint8_t* data, size_t size) {
   RtpHeaderParser::IsRtcp(data, size);
   RtpHeaderParser::GetSsrc(data, size);
-  RTPHeader rtp_header;
-
-  std::unique_ptr<RtpHeaderParser> rtp_header_parser(
-      RtpHeaderParser::CreateForTest());
-
-  rtp_header_parser->Parse(data, size, &rtp_header);
-  for (int i = 1; i < kRtpExtensionNumberOfExtensions; ++i) {
-    if (size > 0 && i >= data[size - 1]) {
-      RTPExtensionType add_extension = static_cast<RTPExtensionType>(i);
-      rtp_header_parser->RegisterRtpHeaderExtension(add_extension, i);
-    }
-  }
-  rtp_header_parser->Parse(data, size, &rtp_header);
-
-  for (int i = 1; i < kRtpExtensionNumberOfExtensions; ++i) {
-    if (size > 1 && i >= data[size - 2]) {
-      RTPExtensionType remove_extension = static_cast<RTPExtensionType>(i);
-      rtp_header_parser->DeregisterRtpHeaderExtension(remove_extension);
-    }
-  }
-  rtp_header_parser->Parse(data, size, &rtp_header);
 }
 
 }  // namespace webrtc
diff --git a/test/fuzzers/utils/BUILD.gn b/test/fuzzers/utils/BUILD.gn
index 6249156058..3e0782f39d 100644
--- a/test/fuzzers/utils/BUILD.gn
+++ b/test/fuzzers/utils/BUILD.gn
@@ -24,6 +24,7 @@ rtc_library("rtp_replayer") {
     "../../../call:call_interfaces",
     "../../../common_video",
     "../../../media:rtc_internal_video_codecs",
+    "../../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:rtc_base_tests_utils",
diff --git a/test/fuzzers/utils/rtp_replayer.cc b/test/fuzzers/utils/rtp_replayer.cc
index a664adb31d..43b1fc2ea4 100644
--- a/test/fuzzers/utils/rtp_replayer.cc
+++ b/test/fuzzers/utils/rtp_replayer.cc
@@ -17,13 +17,13 @@
 
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/transport/field_trial_based_config.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "rtc_base/strings/json.h"
 #include "system_wrappers/include/clock.h"
 #include "test/call_config_utils.h"
 #include "test/encoder_settings.h"
 #include "test/fake_decoder.h"
 #include "test/rtp_file_reader.h"
-#include "test/rtp_header_parser.h"
 #include "test/run_loop.h"
 
 namespace webrtc {
@@ -164,37 +164,32 @@ void RtpReplayer::ReplayPackets(rtc::FakeClock* clock,
           std::min(deliver_in_ms, static_cast<int64_t>(100))));
     }
 
+    rtc::CopyOnWriteBuffer packet_buffer(packet.data, packet.length);
     ++num_packets;
-    switch (call->Receiver()->DeliverPacket(
-        webrtc::MediaType::VIDEO,
-        rtc::CopyOnWriteBuffer(packet.data, packet.length),
-        /* packet_time_us */ -1)) {
+    switch (call->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
+                                            packet_buffer,
+                                            /* packet_time_us */ -1)) {
       case PacketReceiver::DELIVERY_OK:
         break;
       case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
-        RTPHeader header;
-        std::unique_ptr<RtpHeaderParser> parser(
-            RtpHeaderParser::CreateForTest());
-
-        parser->Parse(packet.data, packet.length, &header);
-        if (unknown_packets[header.ssrc] == 0) {
-          RTC_LOG(LS_ERROR) << "Unknown SSRC: " << header.ssrc;
+        webrtc::RtpPacket header;
+        header.Parse(packet_buffer);
+        if (unknown_packets[header.Ssrc()] == 0) {
+          RTC_LOG(LS_ERROR) << "Unknown SSRC: " << header.Ssrc();
         }
-        ++unknown_packets[header.ssrc];
+        ++unknown_packets[header.Ssrc()];
         break;
       }
       case PacketReceiver::DELIVERY_PACKET_ERROR: {
         RTC_LOG(LS_ERROR)
             << "Packet error, corrupt packets or incorrect setup?";
-        RTPHeader header;
-        std::unique_ptr<RtpHeaderParser> parser(
-            RtpHeaderParser::CreateForTest());
-        parser->Parse(packet.data, packet.length, &header);
+        webrtc::RtpPacket header;
+        header.Parse(packet_buffer);
         RTC_LOG(LS_ERROR) << "Packet packet_length=" << packet.length
-                          << " payload_type=" << header.payloadType
-                          << " sequence_number=" << header.sequenceNumber
-                          << " time_stamp=" << header.timestamp
-                          << " ssrc=" << header.ssrc;
+                          << " payload_type=" << header.PayloadType()
+                          << " sequence_number=" << header.SequenceNumber()
+                          << " time_stamp=" << header.Timestamp()
+                          << " ssrc=" << header.Ssrc();
         break;
       }
     }
diff --git a/test/rtp_header_parser.cc b/test/rtp_header_parser.cc
index 45686acb4c..6a7df4b095 100644
--- a/test/rtp_header_parser.cc
+++ b/test/rtp_header_parser.cc
@@ -9,41 +9,10 @@
  */
 #include "test/rtp_header_parser.h"
 
-#include <memory>
-
-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
 
-class RtpHeaderParserImpl : public RtpHeaderParser {
- public:
-  RtpHeaderParserImpl();
-  ~RtpHeaderParserImpl() override = default;
-
-  bool Parse(const uint8_t* packet,
-             size_t length,
-             RTPHeader* header) const override;
-
-  bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
-  bool RegisterRtpHeaderExtension(RtpExtension extension) override;
-
-  bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
-  bool DeregisterRtpHeaderExtension(RtpExtension extension) override;
-
- private:
-  mutable Mutex mutex_;
-  RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(mutex_);
-};
-
-std::unique_ptr<RtpHeaderParser> RtpHeaderParser::CreateForTest() {
-  return std::make_unique<RtpHeaderParserImpl>();
-}
-
-RtpHeaderParserImpl::RtpHeaderParserImpl() {}
-
 bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
   RtpUtility::RtpHeaderParser rtp_parser(packet, length);
   return rtp_parser.RTCP();
@@ -59,43 +28,4 @@ absl::optional<uint32_t> RtpHeaderParser::GetSsrc(const uint8_t* packet,
   return absl::nullopt;
 }
 
-bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
-                                size_t length,
-                                RTPHeader* header) const {
-  RtpUtility::RtpHeaderParser rtp_parser(packet, length);
-  *header = RTPHeader();
-
-  RtpHeaderExtensionMap map;
-  {
-    MutexLock lock(&mutex_);
-    map = rtp_header_extension_map_;
-  }
-
-  const bool valid_rtpheader = rtp_parser.Parse(header, &map);
-  if (!valid_rtpheader) {
-    return false;
-  }
-  return true;
-}
-bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RtpExtension extension) {
-  MutexLock lock(&mutex_);
-  return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri);
-}
-
-bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
-                                                     uint8_t id) {
-  MutexLock lock(&mutex_);
-  return rtp_header_extension_map_.RegisterByType(id, type);
-}
-
-bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) {
-  MutexLock lock(&mutex_);
-  return rtp_header_extension_map_.Deregister(
-      rtp_header_extension_map_.GetType(extension.id));
-}
-
-bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
-  MutexLock lock(&mutex_);
-  return rtp_header_extension_map_.Deregister(type) == 0;
-}
 }  // namespace webrtc
diff --git a/test/rtp_header_parser.h b/test/rtp_header_parser.h
index 851ccf3bc2..ca4eaf657c 100644
--- a/test/rtp_header_parser.h
+++ b/test/rtp_header_parser.h
@@ -10,44 +10,18 @@
 #ifndef TEST_RTP_HEADER_PARSER_H_
 #define TEST_RTP_HEADER_PARSER_H_
 
-#include <memory>
+#include <stddef.h>
+#include <stdint.h>
 
-#include "api/rtp_parameters.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "absl/types/optional.h"
 
 namespace webrtc {
 
-struct RTPHeader;
-
 class RtpHeaderParser {
  public:
-  static std::unique_ptr<RtpHeaderParser> CreateForTest();
-  virtual ~RtpHeaderParser() {}
-
   // Returns true if the packet is an RTCP packet, false otherwise.
   static bool IsRtcp(const uint8_t* packet, size_t length);
   static absl::optional<uint32_t> GetSsrc(const uint8_t* packet, size_t length);
-
-  // Parses the packet and stores the parsed packet in |header|. Returns true on
-  // success, false otherwise.
-  // This method is thread-safe in the sense that it can parse multiple packets
-  // at once.
-  virtual bool Parse(const uint8_t* packet,
-                     size_t length,
-                     RTPHeader* header) const = 0;
-
-  // Registers an RTP header extension and binds it to |id|.
-  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
-                                          uint8_t id) = 0;
-
-  // Registers an RTP header extension.
-  virtual bool RegisterRtpHeaderExtension(RtpExtension extension) = 0;
-
-  // De-registers an RTP header extension.
-  virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
-
-  // De-registers an RTP header extension.
-  virtual bool DeregisterRtpHeaderExtension(RtpExtension extension) = 0;
 };
 }  // namespace webrtc
 #endif  // TEST_RTP_HEADER_PARSER_H_