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
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+++ 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_