115 lines
3.9 KiB
C++
115 lines
3.9 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 MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include "absl/strings/string_view.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
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#include "modules/rtp_rtcp/source/dtmf_queue.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/one_time_event.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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#include "system_wrappers/include/clock.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(absl::string_view payload_name,
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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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bool SendAudio(AudioFrameType frame_type,
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int8_t payload_type,
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uint32_t rtp_timestamp,
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const uint8_t* payload_data,
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size_t payload_size);
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bool SendAudio(AudioFrameType frame_type,
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int8_t payload_type,
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uint32_t rtp_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms);
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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(AudioFrameType 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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Mutex send_audio_mutex_;
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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_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
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uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_mutex_) = 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_ RTC_GUARDED_BY(send_audio_mutex_) = false;
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int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
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int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
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int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
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int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
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int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -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_ RTC_GUARDED_BY(send_audio_mutex_) = 0;
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OneTimeEvent first_packet_sent_;
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absl::optional<uint32_t> encoder_rtp_timestamp_frequency_
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RTC_GUARDED_BY(send_audio_mutex_);
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AbsoluteCaptureTimeSender absolute_capture_time_sender_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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