Move implmentation specific constants out of rtp_header_extension.h

BUG=None

Review-Url: https://codereview.webrtc.org/2642783006
Cr-Commit-Position: refs/heads/master@{#16222}

webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc

diff --git a/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc b/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
index e87a2ed..f1afc1f 100644
--- a/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
+++ b/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
@@ -13,7 +13,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 
 namespace webrtc {
 
@@ -29,8 +29,8 @@
                                        PlayoutDelay playout_delay,
                                        uint16_t seq_num) {
   rtc::CritScope lock(&crit_sect_);
-  RTC_DCHECK_LE(playout_delay.min_ms, kPlayoutDelayMaxMs);
-  RTC_DCHECK_LE(playout_delay.max_ms, kPlayoutDelayMaxMs);
+  RTC_DCHECK_LE(playout_delay.min_ms, PlayoutDelayLimits::kMaxMs);
+  RTC_DCHECK_LE(playout_delay.max_ms, PlayoutDelayLimits::kMaxMs);
   RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
   int64_t unwrapped_seq_num = unwrapper_.Unwrap(seq_num);
   if (playout_delay.min_ms >= 0 &&

webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
index 67e62c2..bbbb143 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
@@ -96,6 +96,7 @@
 }
 
 size_t RtpHeaderExtensionMap::GetTotalLengthInBytes() const {
+  static constexpr size_t kRtpOneByteHeaderLength = 4;
   if (total_values_size_bytes_ == 0)
     return 0;
   return Word32Align(kRtpOneByteHeaderLength + total_values_size_bytes_);

webrtc/modules/rtp_rtcp/source/rtp_header_extension.h

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 777155b..1b5a7ab 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -21,23 +21,6 @@
 
 namespace webrtc {
 
-const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
-
-const size_t kRtpOneByteHeaderLength = 4;
-const size_t kTransmissionTimeOffsetLength = 4;
-const size_t kAudioLevelLength = 2;
-const size_t kAbsoluteSendTimeLength = 4;
-const size_t kVideoRotationLength = 2;
-const size_t kTransportSequenceNumberLength = 3;
-const size_t kPlayoutDelayLength = 4;
-
-// Playout delay in milliseconds. A playout delay limit (min or max)
-// has 12 bits allocated. This allows a range of 0-4095 values which translates
-// to a range of 0-40950 in milliseconds.
-const int kPlayoutDelayGranularityMs = 10;
-// Maximum playout delay value in milliseconds.
-const int kPlayoutDelayMaxMs = 40950;
-
 class RtpHeaderExtensionMap {
  public:
   static constexpr RTPExtensionType kInvalidType = kRtpExtensionNone;

webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index 81f0526..179d279 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -87,6 +87,7 @@
   RtpHeaderExtensionMap map;
   EXPECT_EQ(0u, map.GetTotalLengthInBytes());
   EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+  static constexpr size_t kRtpOneByteHeaderLength = 4;
   EXPECT_EQ(kRtpOneByteHeaderLength + (TransmissionOffset::kValueSizeBytes + 1),
             map.GetTotalLengthInBytes());
 }

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index 098fdc8..def431f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -15,6 +15,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 
 namespace webrtc {
 
@@ -281,6 +282,7 @@
     if (static_cast<size_t>(remain) < (4 + XLen)) {
       return false;
     }
+    static constexpr uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
     if (definedByProfile == kRtpOneByteHeaderExtensionId) {
       const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
       ParseOneByteExtensionHeader(header,
@@ -439,9 +441,9 @@
           int min_playout_delay = (ptr[0] << 4) | ((ptr[1] >> 4) & 0xf);
           int max_playout_delay = ((ptr[1] & 0xf) << 8) | ptr[2];
           header->extension.playout_delay.min_ms =
-              min_playout_delay * kPlayoutDelayGranularityMs;
+              min_playout_delay * PlayoutDelayLimits::kGranularityMs;
           header->extension.playout_delay.max_ms =
-              max_playout_delay * kPlayoutDelayGranularityMs;
+              max_playout_delay * PlayoutDelayLimits::kGranularityMs;
           break;
         }
         case kRtpExtensionNone: