Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 85 insertions, 86 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index dd16fe51b4..1e96d17a67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -19,92 +19,91 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class RTPSenderAudio: public DTMFqueue
-{
-public:
- RTPSenderAudio(Clock* clock,
-                RTPSender* rtpSender,
-                RtpAudioFeedback* audio_feedback);
-    virtual ~RTPSenderAudio();
-
-    int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                 const int8_t payloadType,
-                                 const uint32_t frequency,
-                                 const uint8_t channels,
-                                 const uint32_t rate,
-                                 RtpUtility::Payload*& payload);
-
-    int32_t SendAudio(const FrameType frameType,
-                      const int8_t payloadType,
-                      const uint32_t captureTimeStamp,
-                      const uint8_t* payloadData,
-                      const size_t payloadSize,
-                      const RTPFragmentationHeader* fragmentation);
-
-    // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-    int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
-
-    // Store the audio level in dBov for header-extension-for-audio-level-indication.
-    // Valid range is [0,100]. Actual value is negative.
-    int32_t SetAudioLevel(const uint8_t level_dBov);
-
-    // Send a DTMF tone using RFC 2833 (4733)
-    int32_t SendTelephoneEvent(const uint8_t key,
-                               const uint16_t time_ms,
-                               const uint8_t level);
-
-    int AudioFrequency() const;
-
-    // Set payload type for Redundant Audio Data RFC 2198
-    int32_t SetRED(const int8_t payloadType);
-
-    // Get payload type for Redundant Audio Data RFC 2198
-    int32_t RED(int8_t& payloadType) const;
-
-protected:
-    int32_t SendTelephoneEventPacket(bool ended,
-                                     int8_t dtmf_payload_type,
-                                     uint32_t dtmfTimeStamp,
-                                     uint16_t duration,
-                                     bool markerBit); // set on first packet in talk burst
-
-    bool MarkerBit(const FrameType frameType,
-                   const int8_t payloadType);
-
-private:
- Clock* const _clock;
- RTPSender* const _rtpSender;
- RtpAudioFeedback* const _audioFeedback;
-
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
-
- uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
-
- // DTMF
- bool _dtmfEventIsOn;
- bool _dtmfEventFirstPacketSent;
- int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
- uint32_t _dtmfTimestamp;
- uint8_t _dtmfKey;
- uint32_t _dtmfLengthSamples;
- uint8_t _dtmfLevel;
- int64_t _dtmfTimeLastSent;
- uint32_t _dtmfTimestampLastSent;
-
- int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
-
- // VAD detection, used for markerbit
- bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
- int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
-
- // Audio level indication
- // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
- uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+class RTPSenderAudio : public DTMFqueue {
+ public:
+  RTPSenderAudio(Clock* clock,
+                 RTPSender* rtpSender,
+                 RtpAudioFeedback* audio_feedback);
+  virtual ~RTPSenderAudio();
+
+  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                               int8_t payloadType,
+                               uint32_t frequency,
+                               size_t channels,
+                               uint32_t rate,
+                               RtpUtility::Payload** payload);
+
+  int32_t SendAudio(FrameType frameType,
+                    int8_t payloadType,
+                    uint32_t captureTimeStamp,
+                    const uint8_t* payloadData,
+                    size_t payloadSize,
+                    const RTPFragmentationHeader* fragmentation);
+
+  // set audio packet size, used to determine when it's time to send a DTMF
+  // packet in silence (CNG)
+  int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
+
+  // Store the audio level in dBov for
+  // header-extension-for-audio-level-indication.
+  // Valid range is [0,100]. Actual value is negative.
+  int32_t SetAudioLevel(uint8_t level_dBov);
+
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
+
+  int AudioFrequency() const;
+
+  // Set payload type for Redundant Audio Data RFC 2198
+  int32_t SetRED(int8_t payloadType);
+
+  // Get payload type for Redundant Audio Data RFC 2198
+  int32_t RED(int8_t* payloadType) const;
+
+ protected:
+  int32_t SendTelephoneEventPacket(
+      bool ended,
+      int8_t dtmf_payload_type,
+      uint32_t dtmfTimeStamp,
+      uint16_t duration,
+      bool markerBit);  // set on first packet in talk burst
+
+  bool MarkerBit(const FrameType frameType, const int8_t payloadType);
+
+ private:
+  Clock* const _clock;
+  RTPSender* const _rtpSender;
+  RtpAudioFeedback* const _audioFeedback;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+
+  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
+
+  // DTMF
+  bool _dtmfEventIsOn;
+  bool _dtmfEventFirstPacketSent;
+  int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
+  uint32_t _dtmfTimestamp;
+  uint8_t _dtmfKey;
+  uint32_t _dtmfLengthSamples;
+  uint8_t _dtmfLevel;
+  int64_t _dtmfTimeLastSent;
+  uint32_t _dtmfTimestampLastSent;
+
+  int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
+
+  // VAD detection, used for markerbit
+  bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
+
+  // Audio level indication
+  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_