7c9426cf38
Review URL: https://codereview.webrtc.org/1877253002 Cr-Commit-Position: refs/heads/master@{#12359}
110 lines
3.8 KiB
C++
110 lines
3.8 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/base/criticalsection.h"
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#include "webrtc/base/onetimeevent.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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public:
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RTPSenderAudio(Clock* clock, RTPSender* rtpSender);
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virtual ~RTPSenderAudio();
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int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int8_t payloadType,
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uint32_t frequency,
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size_t channels,
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uint32_t rate,
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RtpUtility::Payload** payload);
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int32_t SendAudio(FrameType frameType,
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int8_t payloadType,
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uint32_t captureTimeStamp,
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const uint8_t* payloadData,
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size_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
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// packet in silence (CNG)
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int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
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// Store the audio level in dBov for
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// 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(uint8_t level_dBov);
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// Send a DTMF tone using RFC 2833 (4733)
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int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
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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(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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protected:
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int32_t SendTelephoneEventPacket(
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bool ended,
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int8_t dtmf_payload_type,
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uint32_t dtmfTimeStamp,
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uint16_t duration,
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bool markerBit); // set on first packet in talk burst
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bool MarkerBit(const FrameType frameType, const int8_t payloadType);
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private:
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Clock* const _clock;
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RTPSender* const _rtpSender;
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rtc::CriticalSection _sendAudioCritsect;
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uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
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// DTMF
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bool _dtmfEventIsOn;
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bool _dtmfEventFirstPacketSent;
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int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
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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 GUARDED_BY(_sendAudioCritsect);
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// VAD detection, used for markerbit
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bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
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int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
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int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
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int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
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int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
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int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
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// Audio level indication
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// (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
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uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
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OneTimeEvent first_packet_sent_;
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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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