Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 174 insertions, 0 deletions

diff --git a/third_party/libwebrtc/moz-patch-stack/0064.patch b/third_party/libwebrtc/moz-patch-stack/0064.patch
new file mode 100644
index 0000000000..34ab2beb03
--- /dev/null
+++ b/third_party/libwebrtc/moz-patch-stack/0064.patch
@@ -0,0 +1,174 @@
+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 a57d9e7f62..de29fd2075 100644
+--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
++++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
+@@ -185,6 +185,7 @@ bool AudioLevel::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
+@@ -237,6 +238,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
+   }
+   return true;
+ }
++#endif
+ 
+ // From RFC 5450: Transmission Time Offsets in RTP Streams.
+ //
+@@ -446,6 +448,7 @@ 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
+@@ -484,6 +487,7 @@ 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 89c73955a2..4b4984bf6d 100644
+--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
++++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
+@@ -88,6 +88,7 @@ class AudioLevel {
+                     uint8_t audio_level);
+ };
+ 
++#if !defined(WEBRTC_MOZILLA_BUILD)
+ class CsrcAudioLevel {
+  public:
+   static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
+@@ -102,6 +103,7 @@ class CsrcAudioLevel {
+   static bool Write(rtc::ArrayView<uint8_t> data,
+                     rtc::ArrayView<const uint8_t> csrc_audio_levels);
+ };
++#endif
+ 
+ class TransmissionOffset {
+  public:
+@@ -292,6 +294,7 @@ 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;
+@@ -306,6 +309,7 @@ class CsrcAudioLevel {
+   static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
+   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 9495841984..fd2f5c5ae8 100644
+--- a/modules/rtp_rtcp/source/rtp_packet.cc
++++ b/modules/rtp_rtcp/source/rtp_packet.cc
+@@ -187,7 +187,9 @@ void RtpPacket::ZeroMutableExtensions() {
+         break;
+       }
+       case RTPExtensionType::kRtpExtensionAudioLevel:
++#if !defined(WEBRTC_MOZILLA_BUILD)
+       case RTPExtensionType::kRtpExtensionCsrcAudioLevel:
++#endif
+       case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
+       case RTPExtensionType::kRtpExtensionColorSpace:
+       case RTPExtensionType::kRtpExtensionGenericFrameDescriptor:
+@@ -205,10 +207,12 @@ 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 0ed7243d0a..336a117f4e 100644
+--- a/modules/rtp_rtcp/source/rtp_sender.cc
++++ b/modules/rtp_rtcp/source/rtp_sender.cc
+@@ -108,7 +108,9 @@ bool IsNonVolatile(RTPExtensionType type) {
+   switch (type) {
+     case kRtpExtensionTransmissionTimeOffset:
+     case kRtpExtensionAudioLevel:
++#if !defined(WEBRTC_MOZILLA_BUILD)
+     case kRtpExtensionCsrcAudioLevel:
++#endif
+     case kRtpExtensionAbsoluteSendTime:
+     case kRtpExtensionTransportSequenceNumber:
+     case kRtpExtensionTransportSequenceNumber02:
+@@ -132,10 +134,12 @@ 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 0e10a8fa3a..5d117529bb 100644
+--- a/test/fuzzers/rtp_packet_fuzzer.cc
++++ b/test/fuzzers/rtp_packet_fuzzer.cc
+@@ -77,11 +77,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
+         uint8_t audio_level;
+         packet.GetExtension<AudioLevel>(&voice_activity, &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);
+@@ -164,11 +166,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
+         // This extension requires state to read and so complicated that
+         // deserves own fuzzer.
+         break;
++#if defined(WEBRTC_MOZILLA_BUILD)
+       case kRtpExtensionCsrcAudioLevel: {
+         CsrcAudioLevelList levels;
+         packet.GetExtension<CsrcAudioLevel>(&levels);
+         break;
+       }
++#endif
+     }
+   }
+ 
+-- 
+2.34.1
+