Merge "Merge upstream-master and enable ARM64" into emu-master-devemu-31-stable-releaseemu-31-release

1 files changed, 9 insertions, 3 deletions

diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index d156352a26..e9ddbb92a1 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -258,6 +258,7 @@ void SetAudioFrameActivityAndType(bool vad_enabled,
 
 int NetEqImpl::GetAudio(AudioFrame* audio_frame,
                         bool* muted,
+                        int* current_sample_rate_hz,
                         absl::optional<Operation> action_override) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   MutexLock lock(&mutex_);
@@ -296,6 +297,11 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame,
     }
   }
 
+  if (current_sample_rate_hz) {
+    *current_sample_rate_hz = delayed_last_output_sample_rate_hz_.value_or(
+        last_output_sample_rate_hz_);
+  }
+
   return kOK;
 }
 
@@ -565,19 +571,19 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
     return kInvalidPointer;
   }
 
-  int64_t receive_time_ms = clock_->TimeInMilliseconds();
+  Timestamp receive_time = clock_->CurrentTime();
   stats_->ReceivedPacket();
 
   PacketList packet_list;
   // Insert packet in a packet list.
-  packet_list.push_back([&rtp_header, &payload, &receive_time_ms] {
+  packet_list.push_back([&rtp_header, &payload, &receive_time] {
     // Convert to Packet.
     Packet packet;
     packet.payload_type = rtp_header.payloadType;
     packet.sequence_number = rtp_header.sequenceNumber;
     packet.timestamp = rtp_header.timestamp;
     packet.payload.SetData(payload.data(), payload.size());
-    packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms);
+    packet.packet_info = RtpPacketInfo(rtp_header, receive_time);
     // Waiting time will be set upon inserting the packet in the buffer.
     RTC_DCHECK(!packet.waiting_time);
     return packet;