diff options
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0058.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0058.patch | 171 |
1 files changed, 171 insertions, 0 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0058.patch b/third_party/libwebrtc/moz-patch-stack/0058.patch new file mode 100644 index 0000000000..00b30f90f9 --- /dev/null +++ b/third_party/libwebrtc/moz-patch-stack/0058.patch @@ -0,0 +1,171 @@ +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 b86a7d775c..ea41226abf 100644 +--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc ++++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc +@@ -181,6 +181,7 @@ 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 +@@ -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 fff7db0294..09bd9912e1 100644 +--- a/modules/rtp_rtcp/source/rtp_header_extensions.h ++++ b/modules/rtp_rtcp/source/rtp_header_extensions.h +@@ -94,6 +94,7 @@ class AudioLevel { + uint8_t audio_level); + }; + ++#if !defined(WEBRTC_MOZILLA_BUILD) + class CsrcAudioLevel { + public: + static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel; +@@ -108,6 +109,7 @@ class CsrcAudioLevel { + static bool Write(rtc::ArrayView<uint8_t> data, + rtc::ArrayView<const uint8_t> csrc_audio_levels); + }; ++#endif + + class TransmissionOffset { + public: +@@ -298,6 +300,7 @@ 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; +@@ -312,6 +315,7 @@ 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 b152cdbd9e..7181b303e1 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 f4ba024444..4972aa0802 100644 +--- a/modules/rtp_rtcp/source/rtp_sender.cc ++++ b/modules/rtp_rtcp/source/rtp_sender.cc +@@ -104,7 +104,9 @@ bool IsNonVolatile(RTPExtensionType type) { + switch (type) { + case kRtpExtensionTransmissionTimeOffset: + case kRtpExtensionAudioLevel: ++#if !defined(WEBRTC_MOZILLA_BUILD) + case kRtpExtensionCsrcAudioLevel: ++#endif + case kRtpExtensionAbsoluteSendTime: + case kRtpExtensionTransportSequenceNumber: + case kRtpExtensionTransportSequenceNumber02: +@@ -128,10 +130,12 @@ 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 + } + } + |