00bceb1eda
The packet size was only used to control how often to output DTMF packets. However, it likely did not work as intended, since that interval was only set during initialization. No changes to the packet size, like what AudioEncoder::Num10MsFramesInNextPacket could indicate, were picked up. The value was instead taken from an entry in ACMCodecDB. Since it was not-fully-functional, its exact value didn't seem to matter and it was getting in the way of making it possible to supply an external audio encoder factory, I've decided to remove it altogether. The DTMF code now uses an interval of 50 ms regardless, which is a value recommended by the RFC. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2545753002 Cr-Commit-Position: refs/heads/master@{#15380}
99 lines
3.5 KiB
C++
99 lines
3.5 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/constructormagic.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 {
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public:
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RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
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~RTPSenderAudio();
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int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_type,
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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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bool SendAudio(FrameType frame_type,
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int8_t payload_type,
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uint32_t capture_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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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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protected:
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bool SendTelephoneEventPacket(
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bool ended,
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uint32_t dtmf_timestamp,
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uint16_t duration,
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bool marker_bit); // set on first packet in talk burst
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bool MarkerBit(FrameType frame_type, int8_t payload_type);
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private:
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Clock* const clock_ = nullptr;
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RTPSender* const rtp_sender_ = nullptr;
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rtc::CriticalSection send_audio_critsect_;
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// DTMF.
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bool dtmf_event_is_on_ = false;
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bool dtmf_event_first_packet_sent_ = false;
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int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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uint32_t dtmf_payload_freq_ GUARDED_BY(send_audio_critsect_) = 8000;
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uint32_t dtmf_timestamp_ = 0;
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uint32_t dtmf_length_samples_ = 0;
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int64_t dtmf_time_last_sent_ = 0;
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uint32_t dtmf_timestamp_last_sent_ = 0;
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DtmfQueue::Event dtmf_current_event_;
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DtmfQueue dtmf_queue_;
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// VAD detection, used for marker bit.
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bool inband_vad_active_ GUARDED_BY(send_audio_critsect_) = false;
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int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_) = -1;
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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 audio_level_dbov_ GUARDED_BY(send_audio_critsect_) = 0;
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OneTimeEvent first_packet_sent_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
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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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