Expose AudioLevel as an absl::optional struct in api/rtp_headers.h

Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.

The old fields are preserved for compatibility with downstream projects, but will be removed in the future.

Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}

api/rtp_headers.cc

@@ -12,6 +12,14 @@
 
 namespace webrtc {
 
+AudioLevel::AudioLevel() : voice_activity_(false), audio_level_(0) {}
+
+AudioLevel::AudioLevel(bool voice_activity, int audio_level)
+    : voice_activity_(voice_activity), audio_level_(audio_level) {
+  RTC_CHECK_GE(audio_level, 0);
+  RTC_CHECK_LE(audio_level, 127);
+}
+
 RTPHeaderExtension::RTPHeaderExtension()
     : hasTransmissionTimeOffset(false),
       transmissionTimeOffset(0),
@@ -34,6 +42,24 @@ RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
 RTPHeaderExtension& RTPHeaderExtension::operator=(
     const RTPHeaderExtension& other) = default;
 
+absl::optional<AudioLevel> RTPHeaderExtension::audio_level() const {
+  if (!hasAudioLevel) {
+    return absl::nullopt;
+  }
+  return AudioLevel(voiceActivity, audioLevel);
+}
+
+void RTPHeaderExtension::set_audio_level(
+    absl::optional<AudioLevel> audio_level) {
+  if (audio_level) {
+    hasAudioLevel = true;
+    voiceActivity = audio_level->voice_activity();
+    audioLevel = audio_level->level();
+  } else {
+    hasAudioLevel = false;
+  }
+}
+
 RTPHeader::RTPHeader()
     : markerBit(false),
       payloadType(0),

api/rtp_headers.h

@@ -77,6 +77,29 @@ struct AbsoluteCaptureTime {
   absl::optional<int64_t> estimated_capture_clock_offset;
 };
 
+// The audio level extension is used to indicate the voice activity and the
+// audio level of the payload in the RTP stream. See:
+// https://tools.ietf.org/html/rfc6464#section-3.
+class AudioLevel {
+ public:
+  AudioLevel();
+  AudioLevel(bool voice_activity, int audio_level);
+  AudioLevel(const AudioLevel& other) = default;
+  AudioLevel& operator=(const AudioLevel& other) = default;
+
+  // Flag indicating whether the encoder believes the audio packet contains
+  // voice activity.
+  bool voice_activity() const { return voice_activity_; }
+
+  // Audio level in -dBov. Values range from 0 to 127, representing 0 to -127
+  // dBov. 127 represents digital silence.
+  int level() const { return audio_level_; }
+
+ private:
+  bool voice_activity_;
+  int audio_level_;
+};
+
 inline bool operator==(const AbsoluteCaptureTime& lhs,
                        const AbsoluteCaptureTime& rhs) {
   return (lhs.absolute_capture_timestamp == rhs.absolute_capture_timestamp) &&
@@ -114,6 +137,12 @@ struct RTPHeaderExtension {
 
   // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
   // https://tools.ietf.org/html/rfc6464#section-3
+  absl::optional<AudioLevel> audio_level() const;
+
+  void set_audio_level(absl::optional<AudioLevel> audio_level);
+
+  // Direct use of the following members is discouraged and will be removed
+  // once downstream projects have been updated.
   bool hasAudioLevel;
   bool voiceActivity;
   uint8_t audioLevel;

api/rtp_packet_info.cc

@@ -37,8 +37,8 @@ RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
 
   csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
 
-  if (extension.hasAudioLevel) {
+  if (extension.audio_level()) {
-    audio_level_ = extension.audioLevel;
+    audio_level_ = extension.audio_level()->level();
   }
 
   absolute_capture_time_ = extension.absolute_capture_time;

audio/audio_send_stream_tests.cc

@@ -116,11 +116,9 @@ TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
       RtpPacket rtp_packet(&extensions_);
       EXPECT_TRUE(rtp_packet.Parse(packet));
 
-      uint8_t audio_level = 0;
+      AudioLevel audio_level;
-      bool voice = false;
+      EXPECT_TRUE(rtp_packet.GetExtension<AudioLevelExtension>(&audio_level));
-      EXPECT_TRUE(
+      if (audio_level.level() != 0) {
-          rtp_packet.GetExtension<AudioLevelExtension>(&voice, &audio_level));
-      if (audio_level != 0) {
         // Wait for at least one packet with a non-zero level.
         observation_complete_.Set();
       } else {

audio/channel_receive_frame_transformer_delegate.cc

@@ -61,7 +61,12 @@ class TransformableIncomingAudioFrame
   const RTPHeader& Header() const { return header_; }
 
   FrameType Type() const override {
-    return header_.extension.voiceActivity ? FrameType::kAudioFrameSpeech
+    if (!header_.extension.audio_level()) {
+      // Audio level extension not set.
+      return FrameType::kAudioFrameCN;
+    }
+    return header_.extension.audio_level()->voice_activity()
+               ? FrameType::kAudioFrameSpeech
                : FrameType::kAudioFrameCN;
   }
 

logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc

@@ -444,16 +444,14 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
   absl::optional<uint64_t> base_audio_level;
   absl::optional<uint64_t> base_voice_activity;
   {
-    bool voice_activity;
+    AudioLevel audio_level;
-    uint8_t audio_level;
+    if (base_event->template GetExtension<AudioLevelExtension>(&audio_level)) {
-    if (base_event->template GetExtension<AudioLevelExtension>(&voice_activity,
+      RTC_DCHECK_LE(audio_level.level(), 0x7Fu);
-                                                               &audio_level)) {
+      base_audio_level = audio_level.level();
-      RTC_DCHECK_LE(audio_level, 0x7Fu);
+      proto_batch->set_audio_level(audio_level.level());
-      base_audio_level = audio_level;
-      proto_batch->set_audio_level(audio_level);
 
-      base_voice_activity = voice_activity;
+      base_voice_activity = audio_level.voice_activity();
-      proto_batch->set_voice_activity(voice_activity);
+      proto_batch->set_voice_activity(audio_level.voice_activity());
     }
   }
 
@@ -641,12 +639,10 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
   // audio_level (RTP extension)
   for (size_t i = 0; i < values.size(); ++i) {
     const EventType* event = batch[i + 1];
-    bool voice_activity;
+    AudioLevel audio_level;
-    uint8_t audio_level;
+    if (event->template GetExtension<AudioLevelExtension>(&audio_level)) {
-    if (event->template GetExtension<AudioLevelExtension>(&voice_activity,
+      RTC_DCHECK_LE(audio_level.level(), 0x7F);
-                                                          &audio_level)) {
+      values[i] = audio_level.level();
-      RTC_DCHECK_LE(audio_level, 0x7Fu);
-      values[i] = audio_level;
     } else {
       values[i].reset();
     }
@@ -659,12 +655,10 @@ void RtcEventLogEncoderNewFormat::EncodeRtpPacket(const Batch& batch,
   // voice_activity (RTP extension)
   for (size_t i = 0; i < values.size(); ++i) {
     const EventType* event = batch[i + 1];
-    bool voice_activity;
+    AudioLevel audio_level;
-    uint8_t audio_level;
+    if (event->template GetExtension<AudioLevelExtension>(&audio_level)) {
-    if (event->template GetExtension<AudioLevelExtension>(&voice_activity,
+      RTC_DCHECK_LE(audio_level.level(), 0x7F);
-                                                          &audio_level)) {
+      values[i] = audio_level.voice_activity();
-      RTC_DCHECK_LE(audio_level, 0x7Fu);
-      values[i] = voice_activity;
     } else {
       values[i].reset();
     }

logging/rtc_event_log/rtc_event_log2rtp_dump.cc

@@ -135,10 +135,9 @@ void ConvertRtpPacket(
   if (incoming.rtp.header.extension.hasTransportSequenceNumber)
     reconstructed_packet.SetExtension<webrtc::TransportSequenceNumber>(
         incoming.rtp.header.extension.transportSequenceNumber);
-  if (incoming.rtp.header.extension.hasAudioLevel)
+  if (incoming.rtp.header.extension.audio_level())
     reconstructed_packet.SetExtension<webrtc::AudioLevelExtension>(
-        incoming.rtp.header.extension.voiceActivity,
+        *incoming.rtp.header.extension.audio_level());
-        incoming.rtp.header.extension.audioLevel);
   if (incoming.rtp.header.extension.hasVideoRotation)
     reconstructed_packet.SetExtension<webrtc::VideoOrientation>(
         incoming.rtp.header.extension.videoRotation);

logging/rtc_event_log/rtc_event_log_parser.cc

@@ -358,13 +358,10 @@ ParsedRtcEventLog::ParseStatus StoreRtpPackets(
     }
     if (proto.has_audio_level()) {
       RTC_PARSE_CHECK_OR_RETURN(proto.has_voice_activity());
-      header.extension.hasAudioLevel = true;
+      bool voice_activity = rtc::checked_cast<bool>(proto.voice_activity());
-      header.extension.voiceActivity =
+      int audio_level = rtc::checked_cast<int>(proto.audio_level());
-          rtc::checked_cast<bool>(proto.voice_activity());
+      RTC_PARSE_CHECK_OR_RETURN_LE(audio_level, 0x7F);
-      const uint8_t audio_level =
+      header.extension.set_audio_level(AudioLevel(voice_activity, audio_level));
-          rtc::checked_cast<uint8_t>(proto.audio_level());
-      RTC_PARSE_CHECK_OR_RETURN_LE(audio_level, 0x7Fu);
-      header.extension.audioLevel = audio_level;
     } else {
       RTC_PARSE_CHECK_OR_RETURN(!proto.has_voice_activity());
     }
@@ -562,13 +559,11 @@ ParsedRtcEventLog::ParseStatus StoreRtpPackets(
     if (audio_level_values.size() > i && audio_level_values[i].has_value()) {
       RTC_PARSE_CHECK_OR_RETURN(voice_activity_values.size() > i &&
                                 voice_activity_values[i].has_value());
-      header.extension.hasAudioLevel = true;
+      bool voice_activity =
-      header.extension.voiceActivity =
           rtc::checked_cast<bool>(voice_activity_values[i].value());
-      const uint8_t audio_level =
+      int audio_level = rtc::checked_cast<int>(audio_level_values[i].value());
-          rtc::checked_cast<uint8_t>(audio_level_values[i].value());
+      RTC_PARSE_CHECK_OR_RETURN_LE(audio_level, 0x7F);
-      RTC_PARSE_CHECK_OR_RETURN_LE(audio_level, 0x7Fu);
+      header.extension.set_audio_level(AudioLevel(voice_activity, audio_level));
-      header.extension.audioLevel = audio_level;
     } else {
       RTC_PARSE_CHECK_OR_RETURN(voice_activity_values.size() <= i ||
                                 !voice_activity_values[i].has_value());

logging/rtc_event_log/rtc_event_log_unittest_helper.cc

@@ -618,8 +618,8 @@ void EventGenerator::RandomizeRtpPacket(
 
   if (extension_map.IsRegistered(AudioLevelExtension::kId) &&
       (all_configured_exts || prng_.Rand<bool>())) {
-    rtp_packet->SetExtension<AudioLevelExtension>(prng_.Rand<bool>(),
+    rtp_packet->SetExtension<AudioLevelExtension>(
-                                                  prng_.Rand(127));
+        AudioLevel(prng_.Rand<bool>(), prng_.Rand(127)));
   }
 
   if (extension_map.IsRegistered(AbsoluteSendTime::kId) &&
@@ -1029,14 +1029,15 @@ void VerifyLoggedRtpHeader(const Event& original_header,
 
   // AudioLevel header extension.
   ASSERT_EQ(original_header.template HasExtension<AudioLevelExtension>(),
-            logged_header.extension.hasAudioLevel);
+            logged_header.extension.audio_level().has_value());
-  if (logged_header.extension.hasAudioLevel) {
+  if (logged_header.extension.audio_level()) {
-    bool voice_activity;
+    AudioLevel audio_level;
-    uint8_t audio_level;
     ASSERT_TRUE(original_header.template GetExtension<AudioLevelExtension>(
-        &voice_activity, &audio_level));
+        &audio_level));
-    EXPECT_EQ(voice_activity, logged_header.extension.voiceActivity);
+    EXPECT_EQ(audio_level.voice_activity(),
-    EXPECT_EQ(audio_level, logged_header.extension.audioLevel);
+              logged_header.extension.audio_level()->voice_activity());
+    EXPECT_EQ(audio_level.level(),
+              logged_header.extension.audio_level()->level());
   }
 
   // VideoOrientation header extension.

media/engine/webrtc_voice_engine_unittest.cc

@@ -1561,7 +1561,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
   webrtc::RtpPacketReceived reference_packet(&extension_map);
   constexpr uint8_t kAudioLevel = 123;
   reference_packet.SetExtension<webrtc::AudioLevelExtension>(
-      /*voice_activity=*/true, kAudioLevel);
+      webrtc::AudioLevel(/*voice_activity=*/true, kAudioLevel));
   //  Create a packet without the extension map but with the same content.
   webrtc::RtpPacketReceived received_packet;
   ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer()));
@@ -1569,12 +1569,10 @@ TEST_P(WebRtcVoiceEngineTestFake, OnPacketReceivedIdentifiesExtensions) {
   receive_channel_->OnPacketReceived(received_packet);
   rtc::Thread::Current()->ProcessMessages(0);
 
-  bool voice_activity;
+  webrtc::AudioLevel audio_level;
-  uint8_t audio_level;
   EXPECT_TRUE(call_.last_received_rtp_packet()
-                  .GetExtension<webrtc::AudioLevelExtension>(&voice_activity,
+                  .GetExtension<webrtc::AudioLevelExtension>(&audio_level));
-                                                             &audio_level));
+  EXPECT_EQ(audio_level.level(), kAudioLevel);
-  EXPECT_EQ(audio_level, kAudioLevel);
 }
 
 // Test that we apply codecs properly.

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -564,8 +564,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   rtp_header.sequenceNumber = 0x1234;
   rtp_header.timestamp = 0x12345678;
   rtp_header.ssrc = 0x87654321;
-  rtp_header.extension.hasAudioLevel = true;
+  rtp_header.extension.set_audio_level(
-  rtp_header.extension.audioLevel = 42;
+      AudioLevel(/*voice_activity=*/false, 42));
 
   EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, SampleRateHz())
@@ -606,7 +606,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
     EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
     EXPECT_THAT(packet_info.csrcs(), IsEmpty());
     EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
-    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.audio_level(),
+              rtp_header.extension.audio_level()->level());
     EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
   }
 
@@ -614,13 +615,13 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   // old, the second one is the expected next packet.
   rtp_header.sequenceNumber -= 1;
   rtp_header.timestamp -= kPayloadLengthSamples;
-  rtp_header.extension.audioLevel = 1;
+  rtp_header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 1));
   payload[0] = 1;
   clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
   rtp_header.sequenceNumber += 2;
   rtp_header.timestamp += 2 * kPayloadLengthSamples;
-  rtp_header.extension.audioLevel = 2;
+  rtp_header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 2));
   payload[0] = 2;
   clock_.AdvanceTimeMilliseconds(2000);
   expected_receive_time = clock_.CurrentTime();
@@ -655,7 +656,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
     EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
     EXPECT_THAT(packet_info.csrcs(), IsEmpty());
     EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
-    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.audio_level(),
+              rtp_header.extension.audio_level()->level());
     EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
   }
 
@@ -772,8 +774,7 @@ TEST_F(NetEqImplTest, InsertRedPayload) {
   AbsoluteCaptureTime capture_time;
   capture_time.absolute_capture_timestamp = 1234;
   header.extension.absolute_capture_time = capture_time;
-  header.extension.audioLevel = 12;
+  header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 12));
-  header.extension.hasAudioLevel = true;
   header.numCSRCs = 1;
   header.arrOfCSRCs[0] = 123;
   neteq_->InsertPacket(header, payload);
@@ -795,7 +796,7 @@ TEST_F(NetEqImplTest, InsertRedPayload) {
   EXPECT_EQ(frame.packet_infos_.size(), 1u);
   EXPECT_EQ(frame.packet_infos_.front().absolute_capture_time(), capture_time);
   EXPECT_EQ(frame.packet_infos_.front().audio_level(),
-            header.extension.audioLevel);
+            header.extension.audio_level()->level());
   EXPECT_EQ(frame.packet_infos_.front().csrcs()[0], header.arrOfCSRCs[0]);
 }
 

modules/audio_coding/neteq/tools/rtp_analyze.cc

@@ -110,9 +110,10 @@ int main(int argc, char* argv[]) {
             static_cast<int>(packet->virtual_packet_length_bytes()),
             packet->header().payloadType, packet->header().markerBit,
             packet->header().ssrc);
-    if (print_audio_level && packet->header().extension.hasAudioLevel) {
+    if (print_audio_level && packet->header().extension.audio_level()) {
-      fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel,
+      fprintf(out_file, " %5d (%1i)",
-              packet->header().extension.voiceActivity);
+              packet->header().extension.audio_level()->level(),
+              packet->header().extension.audio_level()->voice_activity());
     }
     if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
       if (cycles == -1) {

modules/rtp_rtcp/source/rtp_header_extensions.cc

@@ -161,24 +161,41 @@ bool AbsoluteCaptureTimeExtension::Write(rtc::ArrayView<uint8_t> data,
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 // Sample Audio Level Encoding Using the Two-Byte Header Format
 bool AudioLevelExtension::Parse(rtc::ArrayView<const uint8_t> data,
-                                bool* voice_activity,
+                                AudioLevel* extension) {
-                                uint8_t* audio_level) {
   // One-byte and two-byte format share the same data definition.
   if (data.size() != 1)
     return false;
-  *voice_activity = (data[0] & 0x80) != 0;
+  bool voice_activity = (data[0] & 0x80) != 0;
-  *audio_level = data[0] & 0x7F;
+  int audio_level = data[0] & 0x7F;
+  *extension = AudioLevel(voice_activity, audio_level);
+  return true;
+}
+
+bool AudioLevelExtension::Write(rtc::ArrayView<uint8_t> data,
+                                const AudioLevel& extension) {
+  // One-byte and two-byte format share the same data definition.
+  RTC_DCHECK_EQ(data.size(), 1);
+  RTC_CHECK_GE(extension.level(), 0);
+  RTC_CHECK_LE(extension.level(), 0x7f);
+  data[0] = (extension.voice_activity() ? 0x80 : 0x00) | extension.level();
+  return true;
+}
+
+bool AudioLevelExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                bool* voice_activity,
+                                uint8_t* audio_level) {
+  AudioLevel extension;
+  Parse(data, &extension);
+  *voice_activity = extension.voice_activity();
+  *audio_level = extension.level();
   return true;
 }
 
 bool AudioLevelExtension::Write(rtc::ArrayView<uint8_t> data,
                                 bool voice_activity,
                                 uint8_t audio_level) {
-  // One-byte and two-byte format share the same data definition.
+  AudioLevel extension(voice_activity, audio_level);
-  RTC_DCHECK_EQ(data.size(), 1);
+  return Write(data, extension);
-  RTC_CHECK_LE(audio_level, 0x7f);
-  data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
-  return true;
 }
 
 // An RTP Header Extension for Mixer-to-Client Audio Level Indication

modules/rtp_rtcp/source/rtp_header_extensions.h

@@ -84,21 +84,30 @@ class AbsoluteCaptureTimeExtension {
 
 class AudioLevelExtension {
  public:
+  using value_type = AudioLevel;
   static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
   static constexpr uint8_t kValueSizeBytes = 1;
   static constexpr absl::string_view Uri() {
     return RtpExtension::kAudioLevelUri;
   }
 
-  static bool Parse(rtc::ArrayView<const uint8_t> data,
+  static bool Parse(rtc::ArrayView<const uint8_t> data, AudioLevel* extension);
-                    bool* voice_activity,
+  static size_t ValueSize(const AudioLevel& extension) {
-                    uint8_t* audio_level);
-  static size_t ValueSize(bool voice_activity, uint8_t audio_level) {
     return kValueSizeBytes;
   }
-  static bool Write(rtc::ArrayView<uint8_t> data,
+  static bool Write(rtc::ArrayView<uint8_t> data, const AudioLevel& extension);
+
+  [[deprecated("Use AudioLevel struct")]] static bool Parse(
+      rtc::ArrayView<const uint8_t> data,
+      bool* voice_activity,
+      uint8_t* audio_level);
+  [[deprecated("Use AudioLevel struct")]] static size_t ValueSize(
       bool voice_activity,
-                    uint8_t audio_level);
+      uint8_t audio_level) {
+    return kValueSizeBytes;
+  }
+  [[deprecated("Use AudioLevel struct")]] static bool
+  Write(rtc::ArrayView<uint8_t> data, bool voice_activity, uint8_t audio_level);
 };
 
 class CsrcAudioLevel {

modules/rtp_rtcp/source/rtp_packet_received.cc

@@ -61,8 +61,7 @@ void RtpPacketReceived::GetHeader(RTPHeader* header) const {
           &header->extension.feedback_request) ||
       GetExtension<TransportSequenceNumber>(
           &header->extension.transportSequenceNumber);
-  header->extension.hasAudioLevel = GetExtension<AudioLevelExtension>(
+  header->extension.set_audio_level(GetExtension<AudioLevelExtension>());
-      &header->extension.voiceActivity, &header->extension.audioLevel);
   header->extension.hasVideoRotation =
       GetExtension<VideoOrientation>(&header->extension.videoRotation);
   header->extension.hasVideoContentType =

modules/rtp_rtcp/source/rtp_packet_unittest.cc

@@ -237,7 +237,8 @@ TEST(RtpPacketTest, CreateWith2Extensions) {
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
   packet.SetExtension<TransmissionOffset>(kTimeOffset);
-  packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel);
+  packet.SetExtension<AudioLevelExtension>(
+      AudioLevel(kVoiceActive, kAudioLevel));
   EXPECT_THAT(kPacketWithTOAndAL,
               ElementsAreArray(packet.data(), packet.size()));
 }
@@ -257,7 +258,8 @@ TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionFirst) {
                                   TimeDelta::Millis(340));
   ASSERT_TRUE(packet.SetExtension<PlayoutDelayLimits>(playout_delay));
   packet.SetExtension<TransmissionOffset>(kTimeOffset);
-  packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel);
+  packet.SetExtension<AudioLevelExtension>(
+      AudioLevel(kVoiceActive, kAudioLevel));
   EXPECT_THAT(kPacketWithTwoByteExtensionIdFirst,
               ElementsAreArray(packet.data(), packet.size()));
 }
@@ -274,7 +276,8 @@ TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionLast) {
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
   packet.SetExtension<TransmissionOffset>(kTimeOffset);
-  packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel);
+  packet.SetExtension<AudioLevelExtension>(
+      AudioLevel(kVoiceActive, kAudioLevel));
   EXPECT_THAT(kPacketWithTOAndAL,
               ElementsAreArray(packet.data(), packet.size()));
   // Set extension that requires two-byte header.
@@ -334,8 +337,8 @@ TEST(RtpPacketTest, TryToCreateTwoByteHeaderNotSupported) {
   extensions.Register<AudioLevelExtension>(kTwoByteExtensionId);
   RtpPacketToSend packet(&extensions);
   // Set extension that requires two-byte header.
-  EXPECT_FALSE(
+  EXPECT_FALSE(packet.SetExtension<AudioLevelExtension>(
-      packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel));
+      AudioLevel(kVoiceActive, kAudioLevel)));
 }
 
 TEST(RtpPacketTest, CreateTwoByteHeaderSupportedIfExtmapAllowMixed) {
@@ -343,8 +346,8 @@ TEST(RtpPacketTest, CreateTwoByteHeaderSupportedIfExtmapAllowMixed) {
   extensions.Register<AudioLevelExtension>(kTwoByteExtensionId);
   RtpPacketToSend packet(&extensions);
   // Set extension that requires two-byte header.
-  EXPECT_TRUE(
+  EXPECT_TRUE(packet.SetExtension<AudioLevelExtension>(
-      packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel));
+      AudioLevel(kVoiceActive, kAudioLevel)));
 }
 
 TEST(RtpPacketTest, CreateWithMaxSizeHeaderExtension) {
@@ -409,8 +412,8 @@ TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
   EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
   packet.SetPayloadSize(kPayloadSize);
   // Can't set extension after payload.
-  EXPECT_FALSE(
+  EXPECT_FALSE(packet.SetExtension<AudioLevelExtension>(
-      packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel));
+      AudioLevel(kVoiceActive, kAudioLevel)));
   // Unless reserved.
   EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
 }
@@ -688,12 +691,10 @@ TEST(RtpPacketTest, ParseWith2Extensions) {
   int32_t time_offset;
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
-  bool voice_active;
+  AudioLevel audio_level;
-  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevelExtension>(&audio_level));
-  EXPECT_TRUE(
+  EXPECT_EQ(kVoiceActive, audio_level.voice_activity());
-      packet.GetExtension<AudioLevelExtension>(&voice_active, &audio_level));
+  EXPECT_EQ(kAudioLevel, audio_level.level());
-  EXPECT_EQ(kVoiceActive, voice_active);
-  EXPECT_EQ(kAudioLevel, audio_level);
 }
 
 TEST(RtpPacketTest, ParseSecondPacketWithFewerExtensions) {
@@ -721,10 +722,8 @@ TEST(RtpPacketTest, ParseWith2ExtensionsInvalidPadding) {
   int32_t time_offset;
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
-  bool voice_active;
+  AudioLevel audio_level;
-  uint8_t audio_level;
+  EXPECT_FALSE(packet.GetExtension<AudioLevelExtension>(&audio_level));
-  EXPECT_FALSE(
-      packet.GetExtension<AudioLevelExtension>(&voice_active, &audio_level));
 }
 
 TEST(RtpPacketTest, ParseWith2ExtensionsReservedExtensionId) {
@@ -737,10 +736,8 @@ TEST(RtpPacketTest, ParseWith2ExtensionsReservedExtensionId) {
   int32_t time_offset;
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
-  bool voice_active;
+  AudioLevel audio_level;
-  uint8_t audio_level;
+  EXPECT_FALSE(packet.GetExtension<AudioLevelExtension>(&audio_level));
-  EXPECT_FALSE(
-      packet.GetExtension<AudioLevelExtension>(&voice_active, &audio_level));
 }
 
 TEST(RtpPacketTest, ParseWithAllFeatures) {
@@ -792,12 +789,10 @@ TEST(RtpPacketTest, ParseTwoByteHeaderExtensionWithPadding) {
   int32_t time_offset;
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
-  bool voice_active;
+  AudioLevel audio_level;
-  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevelExtension>(&audio_level));
-  EXPECT_TRUE(
+  EXPECT_EQ(kVoiceActive, audio_level.voice_activity());
-      packet.GetExtension<AudioLevelExtension>(&voice_active, &audio_level));
+  EXPECT_EQ(kAudioLevel, audio_level.level());
-  EXPECT_EQ(kVoiceActive, voice_active);
-  EXPECT_EQ(kAudioLevel, audio_level);
 }
 
 TEST(RtpPacketTest, ParseWithExtensionDelayed) {
@@ -1221,7 +1216,8 @@ TEST(RtpPacketTest, RemoveMultipleExtensions) {
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
   packet.SetExtension<TransmissionOffset>(kTimeOffset);
-  packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel);
+  packet.SetExtension<AudioLevelExtension>(
+      AudioLevel(kVoiceActive, kAudioLevel));
 
   EXPECT_THAT(kPacketWithTOAndAL,
               ElementsAreArray(packet.data(), packet.size()));
@@ -1249,7 +1245,8 @@ TEST(RtpPacketTest, RemoveExtensionPreservesOtherUnregisteredExtensions) {
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
   packet.SetExtension<TransmissionOffset>(kTimeOffset);
-  packet.SetExtension<AudioLevelExtension>(kVoiceActive, kAudioLevel);
+  packet.SetExtension<AudioLevelExtension>(
+      AudioLevel(kVoiceActive, kAudioLevel));
 
   EXPECT_THAT(kPacketWithTOAndAL,
               ElementsAreArray(packet.data(), packet.size()));

modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -244,8 +244,8 @@ bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) {
   packet->set_capture_time(clock_->CurrentTime());
   // Set audio level extension, if included.
   packet->SetExtension<AudioLevelExtension>(
-      frame.type == AudioFrameType::kAudioFrameSpeech,
+      AudioLevel(frame.type == AudioFrameType::kAudioFrameSpeech,
-      frame.audio_level_dbov.value_or(127));
+                 frame.audio_level_dbov.value_or(127)));
 
   if (absolute_capture_time.has_value()) {
     // It also checks that extension was registered during SDP negotiation. If

modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc

@@ -121,12 +121,11 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   auto sent_payload = transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
   // Verify AudioLevel extension.
-  bool voice_activity;
+  AudioLevel audio_level;
-  uint8_t audio_level;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevelExtension>(
-      &voice_activity, &audio_level));
+      &audio_level));
-  EXPECT_EQ(kAudioLevel, audio_level);
+  EXPECT_EQ(kAudioLevel, audio_level.level());
-  EXPECT_FALSE(voice_activity);
+  EXPECT_FALSE(audio_level.voice_activity());
 }
 
 TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {

rtc_tools/rtc_event_log_to_text/converter.cc

@@ -333,10 +333,11 @@ bool Convert(std::string inputfile,
       fprintf(output, " transmission_offset=%d",
               event.rtp.header.extension.transmissionTimeOffset);
     }
-    if (event.rtp.header.extension.hasAudioLevel) {
+    if (event.rtp.header.extension.audio_level()) {
       fprintf(output, " voice_activity=%d",
-              event.rtp.header.extension.voiceActivity);
+              event.rtp.header.extension.audio_level()->voice_activity());
-      fprintf(output, " audio_level=%u", event.rtp.header.extension.audioLevel);
+      fprintf(output, " audio_level=%u",
+              event.rtp.header.extension.audio_level()->level());
     }
     if (event.rtp.header.extension.hasVideoRotation) {
       fprintf(output, " video_rotation=%d",
@@ -367,10 +368,11 @@ bool Convert(std::string inputfile,
       fprintf(output, " transmission_offset=%d",
               event.rtp.header.extension.transmissionTimeOffset);
     }
-    if (event.rtp.header.extension.hasAudioLevel) {
+    if (event.rtp.header.extension.audio_level()) {
       fprintf(output, " voice_activity=%d",
-              event.rtp.header.extension.voiceActivity);
+              event.rtp.header.extension.audio_level()->voice_activity());
-      fprintf(output, " audio_level=%u", event.rtp.header.extension.audioLevel);
+      fprintf(output, " audio_level=%u",
+              event.rtp.header.extension.audio_level()->level());
     }
     if (event.rtp.header.extension.hasVideoRotation) {
       fprintf(output, " video_rotation=%d",

rtc_tools/rtc_event_log_visualizer/analyzer.cc

@@ -740,11 +740,12 @@ void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
     TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
                            LineStyle::kLine);
     for (auto& packet : stream.packet_view) {
-      if (packet.header.extension.hasAudioLevel) {
+      if (packet.header.extension.audio_level()) {
         float x = config_.GetCallTimeSec(packet.log_time());
         // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
         // Here we convert it to dBov.
-        float y = static_cast<float>(-packet.header.extension.audioLevel);
+        float y =
+            static_cast<float>(-packet.header.extension.audio_level()->level());
         time_series.points.emplace_back(TimeSeriesPoint(x, y));
       }
     }

test/fuzzers/rtp_packet_fuzzer.cc

@@ -72,11 +72,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
         int32_t offset;
         packet.GetExtension<TransmissionOffset>(&offset);
         break;
-      case kRtpExtensionAudioLevel:
+      case kRtpExtensionAudioLevel: {
-        bool voice_activity;
+        AudioLevel audio_level;
-        uint8_t audio_level;
+        packet.GetExtension<AudioLevelExtension>(&audio_level);
-        packet.GetExtension<AudioLevelExtension>(&voice_activity, &audio_level);
         break;
+      }
       case kRtpExtensionCsrcAudioLevel: {
         std::vector<uint8_t> audio_levels;
         packet.GetExtension<CsrcAudioLevel>(&audio_levels);