Remove usage of webrtc::RTPFragmentationHeader from objc wrappers

Bug: webrtc:6471
Change-Id: Ibe4ce280a9f1aea53016f131d1d235337fe71a4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182502
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32022}
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 7292715..c87dcbe 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -805,8 +805,6 @@
       sources = [
         "objc/api/peerconnection/RTCEncodedImage+Private.h",
         "objc/api/peerconnection/RTCEncodedImage+Private.mm",
-        "objc/api/peerconnection/RTCRtpFragmentationHeader+Private.h",
-        "objc/api/peerconnection/RTCRtpFragmentationHeader+Private.mm",
         "objc/api/peerconnection/RTCVideoCodecInfo+Private.h",
         "objc/api/peerconnection/RTCVideoCodecInfo+Private.mm",
         "objc/api/peerconnection/RTCVideoEncoderSettings+Private.h",
diff --git a/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.h b/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.h
deleted file mode 100644
index 0b0bce5..0000000
--- a/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright 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.
- */
-
-#import "base/RTCRtpFragmentationHeader.h"
-
-#include "modules/include/module_common_types.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-/* Interfaces for converting to/from internal C++ formats. */
-@interface RTC_OBJC_TYPE (RTCRtpFragmentationHeader)
-(Private)
-
-    - (instancetype)initWithNativeFragmentationHeader
-    : (const webrtc::RTPFragmentationHeader *__nullable)fragmentationHeader;
-- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader;
-
-@end
-
-NS_ASSUME_NONNULL_END
diff --git a/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.mm b/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.mm
deleted file mode 100644
index e514cf6..0000000
--- a/sdk/objc/api/peerconnection/RTCRtpFragmentationHeader+Private.mm
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright 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.
- */
-
-#import "RTCRtpFragmentationHeader+Private.h"
-
-#include "modules/include/module_common_types.h"
-
-@implementation RTC_OBJC_TYPE (RTCRtpFragmentationHeader)
-(Private)
-
-    - (instancetype)initWithNativeFragmentationHeader
-    : (const webrtc::RTPFragmentationHeader *)fragmentationHeader {
-  if (self = [super init]) {
-    if (fragmentationHeader) {
-      int count = fragmentationHeader->fragmentationVectorSize;
-      NSMutableArray *offsets = [NSMutableArray array];
-      NSMutableArray *lengths = [NSMutableArray array];
-      NSMutableArray *timeDiffs = [NSMutableArray array];
-      NSMutableArray *plTypes = [NSMutableArray array];
-      for (int i = 0; i < count; ++i) {
-        [offsets addObject:@(fragmentationHeader->fragmentationOffset[i])];
-        [lengths addObject:@(fragmentationHeader->fragmentationLength[i])];
-        [timeDiffs addObject:@(0)];
-        [plTypes addObject:@(0)];
-      }
-      self.fragmentationOffset = [offsets copy];
-      self.fragmentationLength = [lengths copy];
-      self.fragmentationTimeDiff = [timeDiffs copy];
-      self.fragmentationPlType = [plTypes copy];
-    }
-  }
-
-  return self;
-}
-
-- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader {
-  auto fragmentationHeader =
-      std::unique_ptr<webrtc::RTPFragmentationHeader>(new webrtc::RTPFragmentationHeader);
-  fragmentationHeader->VerifyAndAllocateFragmentationHeader(self.fragmentationOffset.count);
-  for (NSUInteger i = 0; i < self.fragmentationOffset.count; ++i) {
-    fragmentationHeader->fragmentationOffset[i] = (size_t)self.fragmentationOffset[i].unsignedIntValue;
-    fragmentationHeader->fragmentationLength[i] = (size_t)self.fragmentationLength[i].unsignedIntValue;
-  }
-
-  return fragmentationHeader;
-}
-
-@end
diff --git a/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm b/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
index 1138064..0aa21f1 100644
--- a/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
+++ b/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
@@ -19,7 +19,6 @@
 #endif
 #import "RTCCodecSpecificInfoH264.h"
 #import "RTCH264ProfileLevelId.h"
-#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
 #import "api/peerconnection/RTCVideoCodecInfo+Private.h"
 #import "base/RTCCodecSpecificInfo.h"
 #import "base/RTCI420Buffer.h"
@@ -32,7 +31,6 @@
 #include "common_video/h264/h264_bitstream_parser.h"
 #include "common_video/h264/profile_level_id.h"
 #include "common_video/include/bitrate_adjuster.h"
-#include "modules/include/module_common_types.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/logging.h"
@@ -783,16 +781,8 @@
   }
 
   __block std::unique_ptr<rtc::Buffer> buffer = std::make_unique<rtc::Buffer>();
-  RTC_OBJC_TYPE(RTCRtpFragmentationHeader) * header;
-  {
-    std::unique_ptr<webrtc::RTPFragmentationHeader> header_cpp;
-    bool result =
-        H264CMSampleBufferToAnnexBBuffer(sampleBuffer, isKeyframe, buffer.get(), &header_cpp);
-    header = [[RTC_OBJC_TYPE(RTCRtpFragmentationHeader) alloc]
-        initWithNativeFragmentationHeader:header_cpp.get()];
-    if (!result) {
-      return;
-    }
+  if (!webrtc::H264CMSampleBufferToAnnexBBuffer(sampleBuffer, isKeyframe, buffer.get())) {
+    return;
   }
 
   RTC_OBJC_TYPE(RTCEncodedImage) *frame = [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] init];
@@ -818,6 +808,8 @@
   _h264BitstreamParser.GetLastSliceQp(&qp);
   frame.qp = @(qp);
 
+  RTC_OBJC_TYPE(RTCRtpFragmentationHeader) *header =
+      [[RTC_OBJC_TYPE(RTCRtpFragmentationHeader) alloc] init];
   BOOL res = _callback(frame, codecSpecificInfo, header);
   if (!res) {
     RTC_LOG(LS_ERROR) << "Encode callback failed";
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.cc b/sdk/objc/components/video_codec/nalu_rewriter.cc
index dc258d6..60382d2 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.cc
+++ b/sdk/objc/components/video_codec/nalu_rewriter.cc
@@ -29,14 +29,10 @@
 const char kAnnexBHeaderBytes[4] = {0, 0, 0, 1};
 const size_t kAvccHeaderByteSize = sizeof(uint32_t);
 
-bool H264CMSampleBufferToAnnexBBuffer(
-    CMSampleBufferRef avcc_sample_buffer,
-    bool is_keyframe,
-    rtc::Buffer* annexb_buffer,
-    std::unique_ptr<RTPFragmentationHeader>* out_header) {
+bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
+                                      bool is_keyframe,
+                                      rtc::Buffer* annexb_buffer) {
   RTC_DCHECK(avcc_sample_buffer);
-  RTC_DCHECK(out_header);
-  out_header->reset(nullptr);
 
   // Get format description from the sample buffer.
   CMVideoFormatDescriptionRef description =
@@ -61,10 +57,6 @@
   // Truncate any previous data in the buffer without changing its capacity.
   annexb_buffer->SetSize(0);
 
-  size_t nalu_offset = 0;
-  std::vector<size_t> frag_offsets;
-  std::vector<size_t> frag_lengths;
-
   // Place all parameter sets at the front of buffer.
   if (is_keyframe) {
     size_t param_set_size = 0;
@@ -80,10 +72,6 @@
       annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
       annexb_buffer->AppendData(reinterpret_cast<const char*>(param_set),
                                 param_set_size);
-      // Update fragmentation.
-      frag_offsets.push_back(nalu_offset + sizeof(kAnnexBHeaderBytes));
-      frag_lengths.push_back(param_set_size);
-      nalu_offset += sizeof(kAnnexBHeaderBytes) + param_set_size;
     }
   }
 
@@ -132,10 +120,6 @@
     // Update buffer.
     annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
     annexb_buffer->AppendData(data_ptr + nalu_header_size, packet_size);
-    // Update fragmentation.
-    frag_offsets.push_back(nalu_offset + sizeof(kAnnexBHeaderBytes));
-    frag_lengths.push_back(packet_size);
-    nalu_offset += sizeof(kAnnexBHeaderBytes) + packet_size;
 
     size_t bytes_written = packet_size + sizeof(kAnnexBHeaderBytes);
     bytes_remaining -= bytes_written;
@@ -143,14 +127,6 @@
   }
   RTC_DCHECK_EQ(bytes_remaining, (size_t)0);
 
-  std::unique_ptr<RTPFragmentationHeader> header(new RTPFragmentationHeader());
-  header->VerifyAndAllocateFragmentationHeader(frag_offsets.size());
-  RTC_DCHECK_EQ(frag_lengths.size(), frag_offsets.size());
-  for (size_t i = 0; i < frag_offsets.size(); ++i) {
-    header->fragmentationOffset[i] = frag_offsets[i];
-    header->fragmentationLength[i] = frag_lengths[i];
-  }
-  *out_header = std::move(header);
   CFRelease(contiguous_buffer);
   return true;
 }
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.h b/sdk/objc/components/video_codec/nalu_rewriter.h
index a0c1aa9..d94ce7b 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.h
+++ b/sdk/objc/components/video_codec/nalu_rewriter.h
@@ -18,7 +18,6 @@
 #include <vector>
 
 #include "common_video/h264/h264_common.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/buffer.h"
 
 using webrtc::H264::NaluIndex;
@@ -27,13 +26,10 @@
 
 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
-// needs to be in Annex B format. Data is written directly to |annexb_buffer|
-// and a new RTPFragmentationHeader is returned in |out_header|.
-bool H264CMSampleBufferToAnnexBBuffer(
-    CMSampleBufferRef avcc_sample_buffer,
-    bool is_keyframe,
-    rtc::Buffer* annexb_buffer,
-    std::unique_ptr<RTPFragmentationHeader>* out_header);
+// needs to be in Annex B format. Data is written directly to |annexb_buffer|.
+bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
+                                      bool is_keyframe,
+                                      rtc::Buffer* annexb_buffer);
 
 // Converts a buffer received from RTP into a sample buffer suitable for the
 // VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
diff --git a/sdk/objc/unittests/nalu_rewriter_xctest.mm b/sdk/objc/unittests/nalu_rewriter_xctest.mm
index 490d228..995cc80 100644
--- a/sdk/objc/unittests/nalu_rewriter_xctest.mm
+++ b/sdk/objc/unittests/nalu_rewriter_xctest.mm
@@ -276,14 +276,12 @@
   // clang-format on
 
   rtc::Buffer annexb_buffer(arraysize(cmsample_data));
-  std::unique_ptr<webrtc::RTPFragmentationHeader> out_header_ptr;
   CMSampleBufferRef sample_buffer =
       [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
 
   Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
                                                             /* is_keyframe */ false,
-                                                            &annexb_buffer,
-                                                            &out_header_ptr);
+                                                            &annexb_buffer);
 
   XCTAssertTrue(result);
 
@@ -293,16 +291,6 @@
       memcmp(expected_annex_b_data, annexb_buffer.data(), arraysize(expected_annex_b_data));
 
   XCTAssertEqual(0, data_comparison_result);
-
-  webrtc::RTPFragmentationHeader* out_header = out_header_ptr.get();
-
-  XCTAssertEqual(2, (int)out_header->Size());
-
-  XCTAssertEqual(4, (int)out_header->Offset(0));
-  XCTAssertEqual(4, (int)out_header->Length(0));
-
-  XCTAssertEqual(12, (int)out_header->Offset(1));
-  XCTAssertEqual(2, (int)out_header->Length(1));
 }
 
 - (void)testH264CMSampleBufferToAnnexBBufferWithKeyframe {
@@ -321,14 +309,12 @@
   // clang-format on
 
   rtc::Buffer annexb_buffer(arraysize(cmsample_data));
-  std::unique_ptr<webrtc::RTPFragmentationHeader> out_header_ptr;
   CMSampleBufferRef sample_buffer =
       [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
 
   Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
                                                             /* is_keyframe */ true,
-                                                            &annexb_buffer,
-                                                            &out_header_ptr);
+                                                            &annexb_buffer);
 
   XCTAssertTrue(result);
 
@@ -341,22 +327,6 @@
                  memcmp(expected_annex_b_data,
                         annexb_buffer.data() + arraysize(SPS_PPS_BUFFER),
                         arraysize(expected_annex_b_data)));
-
-  webrtc::RTPFragmentationHeader* out_header = out_header_ptr.get();
-
-  XCTAssertEqual(4, (int)out_header->Size());
-
-  XCTAssertEqual(4, (int)out_header->Offset(0));
-  XCTAssertEqual(14, (int)out_header->Length(0));
-
-  XCTAssertEqual(22, (int)out_header->Offset(1));
-  XCTAssertEqual(4, (int)out_header->Length(1));
-
-  XCTAssertEqual(30, (int)out_header->Offset(2));
-  XCTAssertEqual(4, (int)out_header->Length(2));
-
-  XCTAssertEqual(38, (int)out_header->Offset(3));
-  XCTAssertEqual(2, (int)out_header->Length(3));
 }
 
 - (CMVideoFormatDescriptionRef)createDescription {