tubestation/third_party/libwebrtc/moz-patch-stack/0056.patch

From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:07:00 +0100
Subject: Bug 1766646 - (fix-ae0d117d51) ifdef our Csrc impl vs upstream's
 impl, see Bug 1771332.

---
 modules/rtp_rtcp/source/rtp_header_extensions.cc | 4 ++++
 modules/rtp_rtcp/source/rtp_header_extensions.h  | 4 ++++
 modules/rtp_rtcp/source/rtp_packet.cc            | 4 ++++
 modules/rtp_rtcp/source/rtp_sender.cc            | 4 ++++
 test/fuzzers/rtp_packet_fuzzer.cc                | 4 ++++
 5 files changed, 20 insertions(+)

diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index bb035729f8..53211fa2b4 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -181,6 +181,7 @@ bool AudioLevelExtension::Write(rtc::ArrayView<uint8_t> data,
   return true;
 }
 
+#if !defined(WEBRTC_MOZILLA_BUILD)
 // An RTP Header Extension for Mixer-to-Client Audio Level Indication
 //
 // https://tools.ietf.org/html/rfc6465
@@ -230,6 +231,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
   }
   return true;
 }
+#endif
 
 // From RFC 5450: Transmission Time Offsets in RTP Streams.
 //
@@ -423,6 +425,7 @@ bool PlayoutDelayLimits::Write(rtc::ArrayView<uint8_t> data,
   return true;
 }
 
+#if defined(WEBRTC_MOZILLA_BUILD)
 // CSRCAudioLevel
 //  Sample Audio Level Encoding Using the One-Byte Header Format
 //  Note that the range of len is 1 to 15 which is encoded as 0 to 14
@@ -461,6 +464,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
   // This extension if used must have at least one audio level
   return csrcAudioLevels.numAudioLevels;
 }
+#endif
 
 // Video Content Type.
 //
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index a42dcfd1cf..134d9cfe4d 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -106,6 +106,7 @@ class AudioLevelExtension {
   static bool Write(rtc::ArrayView<uint8_t> data, const AudioLevel& extension);
 };
 
+#if !defined(WEBRTC_MOZILLA_BUILD)
 class CsrcAudioLevel {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -120,6 +121,7 @@ class CsrcAudioLevel {
   static bool Write(rtc::ArrayView<uint8_t> data,
                     rtc::ArrayView<const uint8_t> csrc_audio_levels);
 };
+#endif
 
 class TransmissionOffset {
  public:
@@ -310,6 +312,7 @@ class ColorSpaceExtension {
   static size_t WriteLuminance(uint8_t* data, float f, int denominator);
 };
 
+#if defined(WEBRTC_MOZILLA_BUILD)
 class CsrcAudioLevel {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -325,6 +328,7 @@ class CsrcAudioLevel {
   static bool Write(rtc::ArrayView<uint8_t> data,
                     const CsrcAudioLevelList& csrcAudioLevels);
 };
+#endif
 
 // Base extension class for RTP header extensions which are strings.
 // Subclasses must defined kId and kUri static constexpr members.
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 2277fb95d7..49b844f71d 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -193,7 +193,9 @@ void RtpPacket::ZeroMutableExtensions() {
         break;
       }
       case RTPExtensionType::kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
       case RTPExtensionType::kRtpExtensionCsrcAudioLevel:
+#endif
       case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
       case RTPExtensionType::kRtpExtensionColorSpace:
       case RTPExtensionType::kRtpExtensionCorruptionDetection:
@@ -212,10 +214,12 @@ void RtpPacket::ZeroMutableExtensions() {
         // Non-mutable extension. Don't change it.
         break;
       }
+#if defined(WEBRTC_MOZILLA_BUILD)
       case RTPExtensionType::kRtpExtensionCsrcAudioLevel: {
         // TODO: This is a Mozilla addition, we need to add a handler for this.
         RTC_CHECK(false);
       }
+#endif
     }
   }
 }
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 1988c61d19..cc8510c8c8 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -115,7 +115,9 @@ bool IsNonVolatile(RTPExtensionType type) {
   switch (type) {
     case kRtpExtensionTransmissionTimeOffset:
     case kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
     case kRtpExtensionCsrcAudioLevel:
+#endif
     case kRtpExtensionAbsoluteSendTime:
     case kRtpExtensionTransportSequenceNumber:
     case kRtpExtensionTransportSequenceNumber02:
@@ -140,10 +142,12 @@ bool IsNonVolatile(RTPExtensionType type) {
     case kRtpExtensionNumberOfExtensions:
       RTC_DCHECK_NOTREACHED();
       return false;
+#if defined(WEBRTC_MOZILLA_BUILD)
     case kRtpExtensionCsrcAudioLevel:
       // TODO: Mozilla implement for CsrcAudioLevel
       RTC_CHECK(false);
       return false;
+#endif
   }
   RTC_CHECK_NOTREACHED();
 }
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 8eeed5cf5a..c9a3899b6f 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -88,11 +88,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
         packet.GetExtension<AudioLevelExtension>(&audio_level);
         break;
       }
+#if !defined(WEBRTC_MOZILLA_BUILD)
       case kRtpExtensionCsrcAudioLevel: {
         std::vector<uint8_t> audio_levels;
         packet.GetExtension<CsrcAudioLevel>(&audio_levels);
         break;
       }
+#endif
       case kRtpExtensionAbsoluteSendTime:
         uint32_t sendtime;
         packet.GetExtension<AbsoluteSendTime>(&sendtime);
@@ -179,11 +181,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
         CorruptionDetectionMessage message;
         packet.GetExtension<CorruptionDetectionExtension>(&message);
         break;
+#if defined(WEBRTC_MOZILLA_BUILD)
       case kRtpExtensionCsrcAudioLevel: {
         CsrcAudioLevelList levels;
         packet.GetExtension<CsrcAudioLevel>(&levels);
         break;
       }
+#endif
     }
   }