Add H264 bitstream rewriting to limit frame reordering marker in header

webrtc/modules/rtp_rtcp/source/h264/h264_common.cc

+/*
+ *  Copyright (c) 2016 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 "webrtc/modules/rtp_rtcp/source/h264/h264_common.h"
+
+namespace webrtc {
+
+const size_t H264Common::kNaluLongStartSequenceSize = 4;
+const size_t H264Common::kNaluShortStartSequenceSize = 3;
+const size_t H264Common::kNaluTypeSize = 1;
+const uint8_t kNaluTypeMask = 0x1F;
+
+std::vector<H264Common::NaluIndex> H264Common::FindNaluIndices(
+    const uint8_t* buffer,
+    size_t buffer_size) {
+  // This is sorta like Boyer-Moore, but with only the first optimization step:
+  // given a 3-byte sequence we're looking at, if the 3rd byte isn't 1 or 0,
+  // skip ahead to the next 3-byte sequence. 0s and 1s are relatively rare, so
+  // this will skip the majority of reads/checks.
+  RTC_CHECK_GE(buffer_size, kNaluShortStartSequenceSize);
+  std::vector<NaluIndex> sequences;
+  const size_t end = buffer_size - kNaluShortStartSequenceSize;
+  for (size_t i = 0; i < end;) {
+    if (buffer[i + 2] > 1) {
+      i += 3;
+    } else if (buffer[i + 2] == 1 && buffer[i + 1] == 0 && buffer[i] == 0) {
+      // We found a start sequence, now check if it was a 3 of 4 byte one.
+      NaluIndex index = {i, i + 3, 0};
+      if (index.start_offset > 0 && buffer[index.start_offset - 1] == 0)
+        --index.start_offset;
+
+      // Update length of previous entry.
+      auto it = sequences.rbegin();
+      if (it != sequences.rend())
+        it->payload_size = index.start_offset - it->payload_start_offset;
+
+      sequences.push_back(index);
+
+      i += 3;
+    } else {
+      ++i;
+    }
+  }
+
+  // Update length of last entry, if any.
+  auto it = sequences.rbegin();
+  if (it != sequences.rend())
+    it->payload_size = buffer_size - it->payload_start_offset;
+
+  return sequences;
+}
+
+H264Common::NaluType webrtc::H264Common::ParseNaluType(uint8_t data) {
+  return static_cast<NaluType>(data & kNaluTypeMask);
+}
+
+std::unique_ptr<rtc::Buffer> webrtc::H264Common::ParseRbsp(const uint8_t* data,
+                                                           size_t length) {
+  // Parse out RBSP, which is basically the source buffer minus emulation
+  // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
+  // section 7.3.1 of the H.264 standard.
+  std::unique_ptr<rtc::Buffer> rbsp_buffer(new rtc::Buffer());
+  const char* sps_bytes = reinterpret_cast<const char*>(data);
+  for (size_t i = 0; i < length;) {
+    // Be careful about over/underflow here. byte_length_ - 3 can underflow, and
+    // i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_
+    // above, and that expression will produce the number of bytes left in
+    // the stream including the byte at i.
+    if (length - i >= 3 && data[i] == 0 && data[i + 1] == 0 &&
+        data[i + 2] == 3) {
+      // Two rbsp bytes + the emulation byte.
+      rbsp_buffer->AppendData(sps_bytes + i, 2);
+      i += 3;
+    } else {
+      // Single rbsp byte.
+      rbsp_buffer->AppendData(sps_bytes[i]);
+      ++i;
+    }
+  }
+  return rbsp_buffer;
+}
+
+// Writes bytes into RBSP, adding emulation (0x03) bytes where necessary.
+void webrtc::H264Common::WriteRbsp(const uint8_t* bytes,
+                                   size_t length,
+                                   rtc::Buffer* destination) {
+  static const uint8_t kZerosInStartSequence = 2;
+  static const uint8_t kEmulationByte = 0x03u;
+  size_t num_consecutive_zeros = 0;
+
+  for (size_t i = 0; i < length; ++i) {
+    uint8_t byte = bytes[i];
+    if (byte <= kEmulationByte &&
+        num_consecutive_zeros >= kZerosInStartSequence) {
+      // Need to escape.
+      destination->AppendData(kEmulationByte);
+      num_consecutive_zeros = 0;
+    }
+    destination->AppendData(byte);
+    if (byte == 0) {
+      ++num_consecutive_zeros;
+    } else {
+      num_consecutive_zeros = 0;
+    }
+  }
+}
+
+}  // namespace webrtc