Bug 1952339 - Vendor libwebrtc from d3cb43adf8

Upstream commit: https://webrtc.googlesource.com/src/+/d3cb43adf8a474152748f7ab910562c38cfe6188
    Move timestamp_aligner.h to webrtc namespace

    Bug: webrtc:42232595
    Change-Id: Iefe0f228f16ef44dd8e45e0074b60f7ccf14f1e1
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378825
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#43984}

Differential Revision: https://phabricator.services.mozilla.com/D243990

third_party/libwebrtc/README.mozilla.last-vendor

@@ -1,4 +1,4 @@
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-03-07T22:34:45.489930+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-03-07T22:35:34.289187+00:00.
 # base of lastest vendoring
-4025b60bc1
+d3cb43adf8

third_party/libwebrtc/modules/audio_device/audio_device_buffer.cc

@@ -264,8 +264,7 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(
 
   if (capture_timestamp_ns) {
     int64_t align_offsync_estimation_time = rtc::TimeMicros();
-    if (align_offsync_estimation_time -
+    if (align_offsync_estimation_time - TimestampAligner::kMinFrameIntervalUs >
-            rtc::TimestampAligner::kMinFrameIntervalUs >
         align_offsync_estimation_time_) {
       align_offsync_estimation_time_ = align_offsync_estimation_time;
       capture_timestamp_ns_ =

third_party/libwebrtc/modules/audio_device/audio_device_buffer.h

@@ -240,7 +240,7 @@ class AudioDeviceBuffer {
 
   // Used for converting capture timestaps (received from AudioRecordThread
   // via AudioRecordJni::DataIsRecorded) to RTC clock.
-  rtc::TimestampAligner timestamp_aligner_;
+  TimestampAligner timestamp_aligner_;
 
 // Should *never* be defined in production builds. Only used for testing.
 // When defined, the output signal will be replaced by a sinus tone at 440Hz.

third_party/libwebrtc/rtc_base/timestamp_aligner.cc

@@ -17,7 +17,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/time_utils.h"
 
-namespace rtc {
+namespace webrtc {
 
 TimestampAligner::TimestampAligner()
     : frames_seen_(0),
@@ -140,4 +140,4 @@ int64_t TimestampAligner::ClipTimestamp(int64_t filtered_time_us,
   return time_us;
 }
 
-}  // namespace rtc
+}  // namespace webrtc

third_party/libwebrtc/rtc_base/timestamp_aligner.h

@@ -16,7 +16,7 @@
 #include "rtc_base/system/rtc_export.h"
 #include "rtc_base/time_utils.h"
 
-namespace rtc {
+namespace webrtc {
 
 // The TimestampAligner class helps translating timestamps of a capture system
 // into the same timescale as is used by rtc::TimeMicros(). Some capture systems
@@ -88,6 +88,12 @@ class RTC_EXPORT TimestampAligner {
   int64_t prev_time_offset_us_;
 };
 
+}  //  namespace webrtc
+
+// Re-export symbols from the webrtc namespace for backwards compatibility.
+// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
+namespace rtc {
+using ::webrtc::TimestampAligner;
 }  // namespace rtc
 
 #endif  // RTC_BASE_TIMESTAMP_ALIGNER_H_

third_party/libwebrtc/rtc_base/timestamp_aligner_unittest.cc

@@ -19,7 +19,7 @@
 #include "rtc_base/time_utils.h"
 #include "test/gtest.h"
 
-namespace rtc {
+namespace webrtc {
 
 namespace {
 // Computes the difference x_k - mean(x), when x_k is the linear sequence x_k =
@@ -60,7 +60,7 @@ void TestTimestampFilter(double rel_freq_error) {
 
   int64_t interval_error_us = kIntervalUs * rel_freq_error;
   int64_t system_start_us = rtc::TimeMicros();
-  webrtc::Random random(17);
+  Random random(17);
 
   int64_t prev_translated_time_us = system_start_us;
 
@@ -204,4 +204,4 @@ TEST(TimestampAlignerTest, TranslateTimestampWithoutStateUpdate) {
   }
 }
 
-}  // namespace rtc
+}  // namespace webrtc