Merging upstream version 126.0.

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

8 files changed, 28 insertions, 14 deletions

diff --git a/third_party/libwebrtc/modules/audio_device/BUILD.gn b/third_party/libwebrtc/modules/audio_device/BUILD.gn
index a135f042db..1672be3f95 100644
--- a/third_party/libwebrtc/modules/audio_device/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_device/BUILD.gn
@@ -91,6 +91,7 @@ if (!build_with_mozilla) { # See Bug 1820869.
     "../../system_wrappers",
     "../../system_wrappers:metrics",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 }
 
@@ -213,7 +214,6 @@ if (!build_with_chromium) {
       "../../rtc_base:platform_thread",
       "../../rtc_base:random",
       "../../rtc_base:rtc_event",
-      "../../rtc_base:rtc_task_queue",
       "../../rtc_base:safe_conversions",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
@@ -461,6 +461,7 @@ rtc_source_set("mock_audio_device") {
     "../../api:make_ref_counted",
     "../../test:test_support",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 }
 
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
index 86240da196..f483b8dc79 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
@@ -13,6 +13,7 @@
 #include <cstdint>
 #include <cstring>
 
+#include "api/array_view.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -107,7 +108,8 @@ void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
 
 void FineAudioBuffer::DeliverRecordedData(
     rtc::ArrayView<const int16_t> audio_buffer,
-    int record_delay_ms) {
+    int record_delay_ms,
+    absl::optional<int64_t> capture_time_ns) {
   RTC_DCHECK(IsReadyForRecord());
   // Always append new data and grow the buffer when needed.
   record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
@@ -118,7 +120,8 @@ void FineAudioBuffer::DeliverRecordedData(
       record_channels_ * record_samples_per_channel_10ms_;
   while (record_buffer_.size() >= num_elements_10ms) {
     audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
-                                            record_samples_per_channel_10ms_);
+                                            record_samples_per_channel_10ms_,
+                                            capture_time_ns);
     audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
     audio_device_buffer_->DeliverRecordedData();
     memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
index a6c3042bb2..7af41d3b21 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
@@ -11,6 +11,10 @@
 #ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 #define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 
+#include <cstddef>
+#include <cstdint>
+
+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
@@ -61,7 +65,12 @@ class FineAudioBuffer {
   // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
   // cache. Call #3 restarts the scheme above.
   void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
-                           int record_delay_ms);
+                           int record_delay_ms) {
+    DeliverRecordedData(audio_buffer, record_delay_ms, absl::nullopt);
+  }
+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
+                           int record_delay_ms,
+                           absl::optional<int64_t> capture_time_ns);
 
  private:
   // Device buffer that works with 10ms chunks of data both for playout and
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 36ea85f7dd..bb9fe63922 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -113,7 +113,7 @@ void RunFineBufferTest(int frame_size_in_samples) {
   {
     InSequence s;
     for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
-      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
+      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms, _))
           .WillOnce(VerifyInputBuffer(j, kChannels * kSamplesPer10Ms))
           .RetiresOnSaturation();
     }
diff --git a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
index 4c29c98f2c..b3923ae67d 100644
--- a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
+++ b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
@@ -21,6 +21,7 @@
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/make_ref_counted.h"
+#include "api/task_queue/task_queue_factory.h"
 #include "common_audio/wav_file.h"
 #include "modules/audio_device/audio_device_impl.h"
 #include "modules/audio_device/include/audio_device_default.h"
@@ -33,7 +34,6 @@
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/random.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/repeating_task.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
diff --git a/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
index b0f54c20ff..0b276185da 100644
--- a/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
+++ b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 #define MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 
+#include "absl/types/optional.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "test/gmock.h"
 
@@ -24,7 +25,9 @@ class MockAudioDeviceBuffer : public AudioDeviceBuffer {
   MOCK_METHOD(int32_t, GetPlayoutData, (void* audioBuffer), (override));
   MOCK_METHOD(int32_t,
               SetRecordedBuffer,
-              (const void* audioBuffer, size_t nSamples),
+              (const void* audioBuffer,
+               size_t nSamples,
+               absl::optional<int64_t> capture_time_ns),
               (override));
   MOCK_METHOD(void, SetVQEData, (int playDelayMS, int recDelayMS), (override));
   MOCK_METHOD(int32_t, DeliverRecordedData, (), (override));
diff --git a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
index 627e68b36f..a3742ea581 100644
--- a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
+++ b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
@@ -19,7 +19,6 @@
 #include "modules/audio_device/include/test_audio_device.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/repeating_task.h"
 
 namespace webrtc {
@@ -59,11 +58,10 @@ TestAudioDevice::TestAudioDevice(
 }
 
 AudioDeviceGeneric::InitStatus TestAudioDevice::Init() {
-  task_queue_ =
-      std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue(
-          "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL));
+  task_queue_ = task_queue_factory_->CreateTaskQueue(
+      "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL);
 
-  RepeatingTaskHandle::Start(task_queue_->Get(), [this]() {
+  RepeatingTaskHandle::Start(task_queue_.get(), [this]() {
     ProcessAudio();
     return TimeDelta::Micros(process_interval_us_);
   });
diff --git a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
index 36192b7f7f..84b48948ba 100644
--- a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
+++ b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "modules/audio_device/audio_device_generic.h"
@@ -22,7 +23,6 @@
 #include "modules/audio_device/include/test_audio_device.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 
@@ -190,7 +190,7 @@ class TestAudioDevice : public AudioDeviceGeneric {
 
   std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
   rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
-  std::unique_ptr<rtc::TaskQueue> task_queue_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
 };
 
 }  // namespace webrtc