Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 110 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
new file mode 100644
index 0000000000..dd16fe51b4
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#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);
+};
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_