-
Notifications
You must be signed in to change notification settings - Fork 1.3k
Commit
This commit does not belong to any branch on this repository, and may belong to a fork outside of the repository.
[WebRTC] Minor improvements for webrtc fuzzers
https://bugs.webkit.org/show_bug.cgi?id=265110 <rdar://118624993> Reviewed by Eric Carlson. * Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc: - Send generated packets through depacketizer. * Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/utils/rtp_replayer.cc: - Only set payload type frequency when it is zero. * Source/ThirdParty/libwebrtc/WebKit/0001-WebRTC-Minor-improvements-for-webrtc-fuzzers.patch: Add. Canonical link: https://commits.webkit.org/271178@main
- Loading branch information
Showing
3 changed files
with
105 additions
and
0 deletions.
There are no files selected for viewing
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
80 changes: 80 additions & 0 deletions
80
Source/ThirdParty/libwebrtc/WebKit/0001-WebRTC-Minor-improvements-for-webrtc-fuzzers.patch
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,80 @@ | ||
| diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc | ||
| index e5550c1279cb..471ae9c3c67d 100644 | ||
| --- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc | ||
| +++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc | ||
| @@ -14,6 +14,9 @@ | ||
| #include "modules/rtp_rtcp/source/rtp_format.h" | ||
| #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" | ||
| #include "modules/rtp_rtcp/source/rtp_packetizer_av1.h" | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| +#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h" | ||
| +#endif | ||
| #include "rtc_base/checks.h" | ||
| #include "test/fuzzers/fuzz_data_helper.h" | ||
|
|
||
| @@ -45,27 +48,42 @@ void FuzzOneInput(const uint8_t* data, size_t size) { | ||
| // When packetization was successful, validate NextPacket function too. | ||
| // While at it, check that packets respect the payload size limits. | ||
| RtpPacketToSend rtp_packet(nullptr); | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + VideoRtpDepacketizerAv1 depacketizer; | ||
| +#endif | ||
| // Single packet. | ||
| if (num_packets == 1) { | ||
| RTC_CHECK(packetizer.NextPacket(&rtp_packet)); | ||
| RTC_CHECK_LE(rtp_packet.payload_size(), | ||
| limits.max_payload_len - limits.single_packet_reduction_len); | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + depacketizer.Parse(rtp_packet.PayloadBuffer()); | ||
| +#endif | ||
| return; | ||
| } | ||
| // First packet. | ||
| RTC_CHECK(packetizer.NextPacket(&rtp_packet)); | ||
| RTC_CHECK_LE(rtp_packet.payload_size(), | ||
| limits.max_payload_len - limits.first_packet_reduction_len); | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + depacketizer.Parse(rtp_packet.PayloadBuffer()); | ||
| +#endif | ||
| // Middle packets. | ||
| for (size_t i = 1; i < num_packets - 1; ++i) { | ||
| RTC_CHECK(packetizer.NextPacket(&rtp_packet)) | ||
| << "Failed to get packet#" << i; | ||
| RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) | ||
| << "Packet #" << i << " exceeds it's limit"; | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + depacketizer.Parse(rtp_packet.PayloadBuffer()); | ||
| +#endif | ||
| } | ||
| // Last packet. | ||
| RTC_CHECK(packetizer.NextPacket(&rtp_packet)); | ||
| RTC_CHECK_LE(rtp_packet.payload_size(), | ||
| limits.max_payload_len - limits.last_packet_reduction_len); | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + depacketizer.Parse(rtp_packet.PayloadBuffer()); | ||
| +#endif | ||
| } | ||
| } // namespace webrtc | ||
| diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/utils/rtp_replayer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/utils/rtp_replayer.cc | ||
| index 83f894dc28db..06a31ab881a2 100644 | ||
| --- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/utils/rtp_replayer.cc | ||
| +++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/utils/rtp_replayer.cc | ||
| @@ -187,8 +187,15 @@ void RtpReplayer::ReplayPackets( | ||
| RTC_LOG(LS_ERROR) << "Packet error, corrupt packets or incorrect setup?"; | ||
| break; | ||
| } | ||
| +#ifdef WEBRTC_WEBKIT_BUILD | ||
| + // Set the clock rate if zero - always 90K for video | ||
| + if (received_packet.payload_type_frequency() == 0) { | ||
| + received_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency); | ||
| + } | ||
| +#else | ||
| // Set the clock rate - always 90K for video | ||
| received_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency); | ||
| +#endif | ||
|
|
||
| call->Receiver()->DeliverRtpPacket( | ||
| MediaType::VIDEO, std::move(received_packet), | ||
| -- | ||
| 2.39.3 (Apple Git-145) | ||
|
|