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tools/webrtc-custom-ios/README.md

@@ -0,0 +1,74 @@
+# Custom WebRTC for Rosetta iOS (Audio E2EE Timestamp)
+
+This setup builds a custom `WebRTC.xcframework` for iOS and patches audio E2EE so
+`FrameEncryptor/FrameDecryptor` receive non-empty `additional_data` with RTP timestamp bytes.
+
+## Why
+
+Stock `stasel/WebRTC` M146 calls audio frame encryptor with empty
+`additional_data` (`ad=0`), so nonce derivation based on timestamp is unavailable.
+
+Desktop uses frame timestamp for nonce. This patch aligns iOS with that approach by passing
+an 8-byte big-endian timestamp payload in `additional_data` (absolute RTP timestamp,
+including sender start offset):
+
+- bytes `0..3` = `0`
+- bytes `4..7` = RTP timestamp (big-endian)
+
+Same patch as Android: `mobile-android/tools/webrtc-custom/patches/0001-...`
+
+## Files
+
+- `build_custom_webrtc_ios.sh` — reproducible build script
+- `patches/0001-audio-e2ee-pass-rtp-timestamp-as-additional-data.patch` — WebRTC patch (identical to Android)
+
+## Build
+
+Requirements: macOS, Xcode, `depot_tools` in PATH.
+
+Bootstrap depot_tools first (if not installed):
+
+```bash
+git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git ~/depot_tools
+export PATH=$PATH:$HOME/depot_tools
+```
+
+Then build:
+
+```bash
+cd /path/to/Rosetta/tools/webrtc-custom-ios
+./build_custom_webrtc_ios.sh
+```
+
+Optional env vars:
+
+- `WEBRTC_ROOT` — checkout root (default: `$HOME/webrtc_ios`)
+- `WEBRTC_BRANCH` — default `branch-heads/6422` (M125, same as Android)
+- `WEBRTC_TAG` — use a specific tag/commit instead of branch
+- `OUT_FRAMEWORK` — output xcframework path (default: `Frameworks/WebRTC.xcframework`)
+- `SYNC_JOBS` — `gclient sync` jobs (default: `1`)
+- `SYNC_RETRIES` — sync retry attempts (default: `8`)
+
+## Integration in Xcode
+
+1. Remove `stasel/WebRTC` SPM dependency from Xcode project
+2. Add `Frameworks/WebRTC.xcframework` as Embedded Framework
+3. Build and verify: diagnostic logs should show `ad=8` instead of `ad=0`
+
+## Verification
+
+After building and integrating:
+
+```
+ENC frame#0 sz=53 ad=8 mode=raw-abs nonce=[00000000XXXXXXXX]
+DEC frame#0 sz=48 ad=8 mode=raw-abs ok=1 nonce=[00000000XXXXXXXX]
+```
+
+Where `XXXXXXXX` is the RTP timestamp in hex. This matches Desktop and Android behavior.
+
+## Maintenance
+
+- Keep patch small: only `audio/channel_send.cc` + `audio/channel_receive.cc`
+- Pin WebRTC branch/tag for releases (M125 = `branch-heads/6422`)
+- Rebuild xcframework on version bumps
+- Verify `ad=8` in diagnostic logs after each rebuild

tools/webrtc-custom-ios/build_custom_webrtc_ios.sh

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+#!/usr/bin/env bash
+set -euo pipefail
+
+# Reproducible custom WebRTC xcframework build for Rosetta iOS.
+# Applies the same E2EE timestamp patch as the Android custom build.
+#
+# Requirements:
+#   - macOS with Xcode (Command Line Tools)
+#   - depot_tools in PATH (run bootstrap_depot_tools.sh first if needed)
+#   - python3, git
+#
+# Usage:
+#   cd /path/to/Rosetta/tools/webrtc-custom-ios
+#   ./build_custom_webrtc_ios.sh
+
+SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
+ROSETTA_DIR="$(cd "${SCRIPT_DIR}/../.." && pwd)"
+PATCH_FILE="${SCRIPT_DIR}/patches/0001-audio-e2ee-pass-rtp-timestamp-as-additional-data.patch"
+
+# Default: WebRTC M125 family — same as Android (branch-heads/6422).
+WEBRTC_BRANCH="${WEBRTC_BRANCH:-branch-heads/6422}"
+WEBRTC_TAG="${WEBRTC_TAG:-}"
+
+# Source checkout root (contains src/).
+WEBRTC_ROOT="${WEBRTC_ROOT:-$HOME/webrtc_ios}"
+WEBRTC_SRC="${WEBRTC_SRC:-${WEBRTC_ROOT}/src}"
+
+# Output framework consumed by Xcode project.
+OUT_FRAMEWORK="${OUT_FRAMEWORK:-${ROSETTA_DIR}/Frameworks/WebRTC.xcframework}"
+
+# Sync tuning (chromium.googlesource rate limits).
+SYNC_JOBS="${SYNC_JOBS:-1}"
+SYNC_RETRIES="${SYNC_RETRIES:-8}"
+SYNC_RETRY_BASE_SEC="${SYNC_RETRY_BASE_SEC:-20}"
+
+echo "[webrtc-ios] root: ${WEBRTC_ROOT}"
+echo "[webrtc-ios] src:  ${WEBRTC_SRC}"
+echo "[webrtc-ios] out:  ${OUT_FRAMEWORK}"
+echo "[webrtc-ios] sync jobs: ${SYNC_JOBS}, retries: ${SYNC_RETRIES}"
+
+export DEPOT_TOOLS_UPDATE=0
+
+retry_cmd() {
+  local max_attempts="$1"
+  shift
+  local attempt=1
+  local backoff="${SYNC_RETRY_BASE_SEC}"
+
+  while true; do
+    if "$@"; then
+      return 0
+    fi
+    if (( attempt >= max_attempts )); then
+      return 1
+    fi
+    echo "[webrtc-ios] attempt ${attempt}/${max_attempts} failed, retrying in ${backoff}s: $*"
+    sleep "${backoff}"
+    backoff=$(( backoff * 2 ))
+    if (( backoff > 300 )); then
+      backoff=300
+    fi
+    attempt=$(( attempt + 1 ))
+  done
+}
+
+sync_with_retry() {
+  local attempt=1
+  while true; do
+    if gclient sync -D --jobs "${SYNC_JOBS}"; then
+      return 0
+    fi
+    if (( attempt >= SYNC_RETRIES )); then
+      echo "[webrtc-ios] ERROR: gclient sync failed after ${SYNC_RETRIES} attempts"
+      return 1
+    fi
+    local wait_sec=$(( SYNC_RETRY_BASE_SEC * attempt ))
+    if (( wait_sec > 300 )); then
+      wait_sec=300
+    fi
+    echo "[webrtc-ios] gclient sync failed (attempt ${attempt}/${SYNC_RETRIES}), sleeping ${wait_sec}s..."
+    sleep "${wait_sec}"
+    attempt=$(( attempt + 1 ))
+  done
+}
+
+if ! command -v fetch >/dev/null 2>&1; then
+  echo "[webrtc-ios] ERROR: depot_tools 'fetch' not found in PATH"
+  echo "[webrtc-ios] Run: git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git"
+  echo "[webrtc-ios] Then: export PATH=\$PATH:/path/to/depot_tools"
+  exit 1
+fi
+
+# Step 1: Checkout WebRTC source for iOS
+if [[ ! -d "${WEBRTC_SRC}/.git" ]]; then
+  echo "[webrtc-ios] checkout not found, fetching webrtc_ios..."
+  mkdir -p "${WEBRTC_ROOT}"
+  pushd "${WEBRTC_ROOT}" >/dev/null
+  retry_cmd "${SYNC_RETRIES}" fetch --nohooks --no-history webrtc_ios
+  sync_with_retry
+  popd >/dev/null
+fi
+
+pushd "${WEBRTC_SRC}" >/dev/null
+
+# Step 2: Sync to target branch
+echo "[webrtc-ios] syncing source..."
+retry_cmd "${SYNC_RETRIES}" git fetch --all --tags
+
+if [[ -n "${WEBRTC_TAG}" ]]; then
+  retry_cmd "${SYNC_RETRIES}" git checkout "${WEBRTC_TAG}"
+else
+  if git show-ref --verify --quiet "refs/remotes/origin/${WEBRTC_BRANCH}"; then
+    retry_cmd "${SYNC_RETRIES}" git checkout -B "${WEBRTC_BRANCH}" "origin/${WEBRTC_BRANCH}"
+  else
+    retry_cmd "${SYNC_RETRIES}" git checkout "${WEBRTC_BRANCH}"
+  fi
+  if git rev-parse --abbrev-ref --symbolic-full-name '@{u}' >/dev/null 2>&1; then
+    retry_cmd "${SYNC_RETRIES}" git pull --ff-only
+  fi
+fi
+
+sync_with_retry
+
+# Step 3: Apply E2EE patch
+echo "[webrtc-ios] applying Rosetta E2EE patch..."
+git reset --hard
+echo "[webrtc-ios] apply $(basename "${PATCH_FILE}")"
+git apply --check "${PATCH_FILE}"
+git apply "${PATCH_FILE}"
+echo "[webrtc-ios] patch applied successfully"
+
+# Step 4: Build iOS framework
+echo "[webrtc-ios] building xcframework (this takes 30-60 minutes)..."
+mkdir -p "$(dirname "${OUT_FRAMEWORK}")"
+
+# Remove previous build if exists
+rm -rf "${OUT_FRAMEWORK}"
+
+python3 tools_webrtc/ios/build_ios_libs.py \
+  --build-dir out_rosetta_ios \
+  --output-dir "$(dirname "${OUT_FRAMEWORK}")" \
+  --arch arm64 x86_64 \
+  --extra-gn-args "is_debug=false is_component_build=false rtc_include_tests=false rtc_build_examples=false"
+
+echo "[webrtc-ios] done"
+echo "[webrtc-ios] framework: ${OUT_FRAMEWORK}"
+echo ""
+echo "Next steps:"
+echo "  1. Remove stasel/WebRTC SPM dependency from Xcode"
+echo "  2. Add ${OUT_FRAMEWORK} as Embedded Framework"
+echo "  3. Build and verify: logs should show ad=8 (not ad=0)"
+
+popd >/dev/null

tools/webrtc-custom-ios/patches/0001-audio-e2ee-pass-rtp-timestamp-as-additional-data.patch

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+diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
+index 17cf859ed8..b9d9ab14c8 100644
+--- a/audio/channel_receive.cc
++++ b/audio/channel_receive.cc
+@@ -693,10 +693,20 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
+ 
+     const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
+                                       header.arrOfCSRCs + header.numCSRCs);
++    const uint8_t additional_data_bytes[8] = {
++        0,
++        0,
++        0,
++        0,
++        static_cast<uint8_t>((header.timestamp >> 24) & 0xff),
++        static_cast<uint8_t>((header.timestamp >> 16) & 0xff),
++        static_cast<uint8_t>((header.timestamp >> 8) & 0xff),
++        static_cast<uint8_t>(header.timestamp & 0xff),
++    };
+     const FrameDecryptorInterface::Result decrypt_result =
+         frame_decryptor_->Decrypt(
+             cricket::MEDIA_TYPE_AUDIO, csrcs,
+-            /*additional_data=*/nullptr,
++            /*additional_data=*/additional_data_bytes,
+             rtc::ArrayView<const uint8_t>(payload, payload_data_length),
+             decrypted_audio_payload);
+ 
+diff --git a/audio/channel_send.cc b/audio/channel_send.cc
+index 4a2700177b..7ebb501704 100644
+--- a/audio/channel_send.cc
++++ b/audio/channel_send.cc
+@@ -320,10 +320,23 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
+ 
+       // Encrypt the audio payload into the buffer.
+       size_t bytes_written = 0;
++      const uint32_t additional_data_timestamp =
++          rtp_timestamp_without_offset + rtp_rtcp_->StartTimestamp();
++      const uint8_t additional_data_bytes[8] = {
++          0,
++          0,
++          0,
++          0,
++          static_cast<uint8_t>((additional_data_timestamp >> 24) & 0xff),
++          static_cast<uint8_t>((additional_data_timestamp >> 16) & 0xff),
++          static_cast<uint8_t>((additional_data_timestamp >> 8) & 0xff),
++          static_cast<uint8_t>(additional_data_timestamp & 0xff),
++      };
++
+       int encrypt_status = frame_encryptor_->Encrypt(
+           cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
+-          /*additional_data=*/nullptr, payload, encrypted_audio_payload,
+-          &bytes_written);
++          /*additional_data=*/additional_data_bytes, payload,
++          encrypted_audio_payload, &bytes_written);
+       if (encrypt_status != 0) {
+         RTC_DLOG(LS_ERROR)
+             << "Channel::SendData() failed encrypt audio payload: "