ebdb0e3ad0
- Add ability to VoE to send Absolute Sender Time header extension. - Refactor handling of RTP header extensions in VoE to work the same as in ViE. - Add API to enable receiving Absolute Sender Time in VoE. This is part of the work to include audio packets in bandwidth estimation, for better accuracy in estimates. BUG= TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
118 lines
4.1 KiB
C++
118 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPSenderAudio: public DTMFqueue
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{
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public:
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RTPSenderAudio(const int32_t id, Clock* clock,
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RTPSender* rtpSender);
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virtual ~RTPSenderAudio();
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int32_t RegisterAudioPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payloadType,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate,
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ModuleRTPUtility::Payload*& payload);
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int32_t SendAudio(const FrameType frameType,
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const int8_t payloadType,
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const uint32_t captureTimeStamp,
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const uint8_t* payloadData,
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const uint32_t payloadSize,
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const RTPFragmentationHeader* fragmentation);
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// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
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int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
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// Store the audio level in dBov for header-extension-for-audio-level-indication.
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// Valid range is [0,100]. Actual value is negative.
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int32_t SetAudioLevel(const uint8_t level_dBov);
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// Send a DTMF tone using RFC 2833 (4733)
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int32_t SendTelephoneEvent(const uint8_t key,
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const uint16_t time_ms,
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const uint8_t level);
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bool SendTelephoneEventActive(int8_t& telephoneEvent) const;
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void SetAudioFrequency(const uint32_t f);
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int AudioFrequency() const;
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// Set payload type for Redundant Audio Data RFC 2198
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int32_t SetRED(const int8_t payloadType);
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// Get payload type for Redundant Audio Data RFC 2198
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int32_t RED(int8_t& payloadType) const;
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int32_t RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
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protected:
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int32_t SendTelephoneEventPacket(const bool ended,
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const uint32_t dtmfTimeStamp,
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const uint16_t duration,
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const bool markerBit); // set on first packet in talk burst
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bool MarkerBit(const FrameType frameType,
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const int8_t payloadType);
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private:
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int32_t _id;
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Clock* _clock;
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RTPSender* _rtpSender;
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CriticalSectionWrapper* _audioFeedbackCritsect;
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RtpAudioFeedback* _audioFeedback;
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CriticalSectionWrapper* _sendAudioCritsect;
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uint32_t _frequency;
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uint16_t _packetSizeSamples;
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// DTMF
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bool _dtmfEventIsOn;
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bool _dtmfEventFirstPacketSent;
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int8_t _dtmfPayloadType;
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uint32_t _dtmfTimestamp;
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uint8_t _dtmfKey;
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uint32_t _dtmfLengthSamples;
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uint8_t _dtmfLevel;
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int64_t _dtmfTimeLastSent;
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uint32_t _dtmfTimestampLastSent;
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int8_t _REDPayloadType;
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// VAD detection, used for markerbit
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bool _inbandVADactive;
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int8_t _cngNBPayloadType;
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int8_t _cngWBPayloadType;
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int8_t _cngSWBPayloadType;
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int8_t _cngFBPayloadType;
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int8_t _lastPayloadType;
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// Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
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uint8_t _audioLevel_dBov;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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