4c556219e5
Combine all parameters into single struct so that it is easier to add and remove optional parameters Use Timestamp type instad of plain int to represent capture time Use rtc::ArrayView instead of pointer+size to represent payload Merge passing audio level into send function. Bug: webrtc:13757, webrtc:14870 Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40688}
139 lines
4.9 KiB
C++
139 lines
4.9 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/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() = delete;
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RTPSenderAudio(const RTPSenderAudio&) = delete;
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RTPSenderAudio& operator=(const RTPSenderAudio&) = delete;
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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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struct RtpAudioFrame {
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AudioFrameType type = AudioFrameType::kAudioFrameSpeech;
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rtc::ArrayView<const uint8_t> payload;
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// Payload id to write to the payload type field of the rtp packet.
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int payload_id = -1;
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// capture time of the audio frame represented as rtp timestamp.
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uint32_t rtp_timestamp = 0;
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// capture time of the audio frame in the same epoch as `clock->CurrentTime`
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absl::optional<Timestamp> capture_time;
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// Audio level in dBov for
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// header-extension-for-audio-level-indication.
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// Valid range is [0,127]. Actual value is negative.
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absl::optional<int> audio_level_dbov;
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};
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bool SendAudio(const RtpAudioFrame& frame);
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[[deprecated]] 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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// `absolute_capture_timestamp_ms` and `Clock::CurrentTime`
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// should be using the same epoch.
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[[deprecated]] 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 [0,127]. Actual value is negative.
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[[deprecated]] 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_) = 127;
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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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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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