Delete RtpDepacketizer::Create factory

Bug: webrtc:11152
Change-Id: I09824b97506a11f917cd71f2f0d30306538eee13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163023
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30178}

modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc

@@ -14,7 +14,8 @@
 
 #include "absl/memory/memory.h"
 #include "absl/types/optional.h"
-#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_depacketizer_av1.h"
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
@@ -28,17 +29,14 @@ namespace {
 // Wrapper over legacy RtpDepacketizer interface.
 // TODO(bugs.webrtc.org/11152): Delete when all RtpDepacketizers updated to
 // the VideoRtpDepacketizer interface.
-class LegacyRtpDepacketizer : public VideoRtpDepacketizer {
+template <typename Depacketizer>
+class Legacy : public VideoRtpDepacketizer {
  public:
-  explicit LegacyRtpDepacketizer(VideoCodecType codec) : codec_(codec) {}
-  ~LegacyRtpDepacketizer() override = default;
-
   absl::optional<ParsedRtpPayload> Parse(
       rtc::CopyOnWriteBuffer rtp_payload) override {
-    auto depacketizer = absl::WrapUnique(RtpDepacketizer::Create(codec_));
+    Depacketizer depacketizer;
-    RTC_CHECK(depacketizer);
     RtpDepacketizer::ParsedPayload parsed_payload;
-    if (!depacketizer->Parse(&parsed_payload, rtp_payload.cdata(),
+    if (!depacketizer.Parse(&parsed_payload, rtp_payload.cdata(),
                             rtp_payload.size())) {
       return absl::nullopt;
     }
@@ -48,9 +46,6 @@ class LegacyRtpDepacketizer : public VideoRtpDepacketizer {
                                   parsed_payload.payload_length);
     return result;
   }
-
- private:
-  const VideoCodecType codec_;
 };
 
 }  // namespace
@@ -59,13 +54,13 @@ std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
     VideoCodecType codec) {
   switch (codec) {
     case kVideoCodecH264:
-      return std::make_unique<LegacyRtpDepacketizer>(codec);
+      return std::make_unique<Legacy<RtpDepacketizerH264>>();
     case kVideoCodecVP8:
       return std::make_unique<VideoRtpDepacketizerVp8>();
     case kVideoCodecVP9:
       return std::make_unique<VideoRtpDepacketizerVp9>();
     case kVideoCodecAV1:
-      return std::make_unique<LegacyRtpDepacketizer>(codec);
+      return std::make_unique<Legacy<RtpDepacketizerAv1>>();
     case kVideoCodecGeneric:
     case kVideoCodecMultiplex:
       return std::make_unique<VideoRtpDepacketizerGeneric>();

modules/rtp_rtcp/source/rtp_format.cc

@@ -13,7 +13,6 @@
 #include <memory>
 
 #include "absl/types/variant.h"
-#include "modules/rtp_rtcp/source/rtp_depacketizer_av1.h"
 #include "modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
@@ -143,24 +142,4 @@ std::vector<int> RtpPacketizer::SplitAboutEqually(
   return result;
 }
 
-RtpDepacketizer* RtpDepacketizer::Create(absl::optional<VideoCodecType> type) {
-  if (!type) {
-    // Use raw depacketizer.
-    return new RtpDepacketizerGeneric(/*generic_header_enabled=*/false);
-  }
-
-  switch (*type) {
-    case kVideoCodecH264:
-      return new RtpDepacketizerH264();
-    case kVideoCodecVP8:
-      return new RtpDepacketizerVp8();
-    case kVideoCodecVP9:
-      return new RtpDepacketizerVp9();
-    case kVideoCodecAV1:
-      return new RtpDepacketizerAv1();
-    default:
-      return new RtpDepacketizerGeneric(/*generic_header_enabled=*/true);
-  }
-}
-
 }  // namespace webrtc

modules/rtp_rtcp/source/rtp_format.h

@@ -76,9 +76,6 @@ class RtpDepacketizer {
     size_t payload_length;
   };
 
-  // If type is not set, returns a raw depacketizer.
-  static RtpDepacketizer* Create(absl::optional<VideoCodecType> type);
-
   virtual ~RtpDepacketizer() {}
 
   // Parses the RTP payload, parsed result will be saved in |parsed_payload|.

modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

@@ -514,7 +514,7 @@ struct H264ParsedPayload : public RtpDepacketizer::ParsedPayload {
 class RtpDepacketizerH264Test : public ::testing::Test {
  protected:
   RtpDepacketizerH264Test()
-      : depacketizer_(RtpDepacketizer::Create(kVideoCodecH264)) {}
+      : depacketizer_(std::make_unique<RtpDepacketizerH264>()) {}
 
   void ExpectPacket(H264ParsedPayload* parsed_payload,
                     const uint8_t* data,

modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc

@@ -174,7 +174,7 @@ TEST(RtpPacketizerVp8Test, TIDAndKeyIdx) {
 class RtpDepacketizerVp8Test : public ::testing::Test {
  protected:
   RtpDepacketizerVp8Test()
-      : depacketizer_(RtpDepacketizer::Create(kVideoCodecVP8)) {}
+      : depacketizer_(std::make_unique<RtpDepacketizerVp8>()) {}
 
   void ExpectPacket(RtpDepacketizer::ParsedPayload* parsed_payload,
                     const uint8_t* data,