aa3c1cc927
The replacements are absl::EqualsIgnoreCase and absl::StartsWithIgnoreCase. Also delete the alias RtpUtility::StringCompare. Bug: webrtc:6424 Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f Reviewed-on: https://webrtc-review.googlesource.com/c/109006 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25481}
101 lines
3.5 KiB
C++
101 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 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 "common_types.h" // NOLINT(build/include)
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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 "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/onetimeevent.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(const char* payloadName,
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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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// 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_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
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uint32_t dtmf_payload_freq_ RTC_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_ RTC_GUARDED_BY(send_audio_critsect_) = false;
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int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
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int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
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int8_t last_payload_type_ RTC_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_ RTC_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 // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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