7c4d20fd6c
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls by doing a lot shorter locking which fetches a current state of RTPSenderAudio variables before sending. Thread annotates locked variables and removes one lock in RTPSenderAudio, bonus fixes data races reported in voe_auto_test --automated under TSan (DTMF data race). Also includes some bonus cleanup of RTPSenderVideo which removes the send critsect completely as all methods using it was always called from RTPSender under its send_critsect. R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org BUG=3001, chromium:454654 Review URL: https://webrtc-codereview.appspot.com/41869004 Cr-Commit-Position: refs/heads/master@{#8348} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
113 lines
4.1 KiB
C++
113 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,
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Clock* clock,
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RTPSender* rtpSender,
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RtpAudioFeedback* audio_feedback);
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virtual ~RTPSenderAudio();
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int32_t RegisterAudioPayload(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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RtpUtility::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 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 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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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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protected:
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int32_t SendTelephoneEventPacket(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,
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const int8_t payloadType);
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private:
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const int32_t _id;
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Clock* const _clock;
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RTPSender* const _rtpSender;
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RtpAudioFeedback* const _audioFeedback;
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scoped_ptr<CriticalSectionWrapper> _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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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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