Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 80 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
new file mode 100644
index 0000000000..53ef2a61f4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+
+#include <stdint.h>
+#include <stddef.h>
+
+namespace rtc {
+class BitBuffer;
+}
+
+namespace webrtc {
+
+// Stateful H264 bitstream parser (due to SPS/PPS). Used to parse out QP values
+// from the bitstream.
+// TODO(pbos): Unify with RTP SPS parsing and only use one H264 parser.
+// TODO(pbos): If/when this gets used on the receiver side CHECKs must be
+// removed and gracefully abort as we have no control over receive-side
+// bitstreams.
+class H264BitstreamParser {
+ public:
+  // Parse an additional chunk of H264 bitstream.
+  void ParseBitstream(const uint8_t* bitstream, size_t length);
+
+  // Get the last extracted QP value from the parsed bitstream.
+  bool GetLastSliceQp(int* qp) const;
+
+ private:
+  // Captured in SPS and used when parsing slice NALUs.
+  struct SpsState {
+    SpsState();
+
+    uint32_t delta_pic_order_always_zero_flag = 0;
+    uint32_t separate_colour_plane_flag = 0;
+    uint32_t frame_mbs_only_flag = 0;
+    uint32_t log2_max_frame_num_minus4 = 0;
+    uint32_t log2_max_pic_order_cnt_lsb_minus4 = 0;
+    uint32_t pic_order_cnt_type = 0;
+  };
+
+  struct PpsState {
+    PpsState();
+
+    bool bottom_field_pic_order_in_frame_present_flag = false;
+    bool weighted_pred_flag = false;
+    uint32_t weighted_bipred_idc = false;
+    uint32_t redundant_pic_cnt_present_flag = 0;
+    int pic_init_qp_minus26 = 0;
+  };
+
+  void ParseSlice(const uint8_t* slice, size_t length);
+  bool ParseSpsNalu(const uint8_t* sps_nalu, size_t length);
+  bool ParsePpsNalu(const uint8_t* pps_nalu, size_t length);
+  bool ParseNonParameterSetNalu(const uint8_t* source,
+                                size_t source_length,
+                                uint8_t nalu_type);
+
+  // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
+  bool sps_parsed_ = false;
+  SpsState sps_;
+  bool pps_parsed_ = false;
+  PpsState pps_;
+
+  // Last parsed slice QP.
+  bool last_slice_qp_delta_parsed_ = false;
+  int32_t last_slice_qp_delta_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_