| /* | |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ | |
| #define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ | |
| #include <stddef.h> | |
| #include <stdint.h> | |
| #include <optional> | |
| #include "absl/strings/string_view.h" | |
| #include "api/array_view.h" | |
| #include "api/units/timestamp.h" | |
| #include "modules/audio_coding/include/audio_coding_module_typedefs.h" | |
| #include "modules/rtp_rtcp/source/absolute_capture_time_sender.h" | |
| #include "modules/rtp_rtcp/source/dtmf_queue.h" | |
| #include "modules/rtp_rtcp/source/rtp_sender.h" | |
| #include "rtc_base/one_time_event.h" | |
| #include "rtc_base/synchronization/mutex.h" | |
| #include "rtc_base/thread_annotations.h" | |
| #include "system_wrappers/include/clock.h" | |
| namespace webrtc { | |
| class RTPSenderAudio { | |
| public: | |
| RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); | |
| RTPSenderAudio() = delete; | |
| RTPSenderAudio(const RTPSenderAudio&) = delete; | |
| RTPSenderAudio& operator=(const RTPSenderAudio&) = delete; | |
| ~RTPSenderAudio(); | |
| int32_t RegisterAudioPayload(absl::string_view payload_name, | |
| int8_t payload_type, | |
| uint32_t frequency, | |
| size_t channels, | |
| uint32_t rate); | |
| struct RtpAudioFrame { | |
| AudioFrameType type = AudioFrameType::kAudioFrameSpeech; | |
| ArrayView<const uint8_t> payload; | |
| // Payload id to write to the payload type field of the rtp packet. | |
| int payload_id = -1; | |
| // capture time of the audio frame represented as rtp timestamp. | |
| uint32_t rtp_timestamp = 0; | |
| // capture time of the audio frame in the same epoch as `clock->CurrentTime` | |
| std::optional<Timestamp> capture_time; | |
| // Audio level in dBov for | |
| // header-extension-for-audio-level-indication. | |
| // Valid range is [0,127]. Actual value is negative. | |
| std::optional<int> audio_level_dbov; | |
| // Contributing sources list. | |
| ArrayView<const uint32_t> csrcs; | |
| }; | |
| bool SendAudio(const RtpAudioFrame& frame); | |
| // Send a DTMF tone using RFC 2833 (4733) | |
| int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); | |
| protected: | |
| bool SendTelephoneEventPacket( | |
| bool ended, | |
| uint32_t dtmf_timestamp, | |
| uint16_t duration, | |
| bool marker_bit); // set on first packet in talk burst | |
| bool MarkerBit(AudioFrameType frame_type, int8_t payload_type); | |
| private: | |
| Clock* const clock_ = nullptr; | |
| RTPSender* const rtp_sender_ = nullptr; | |
| Mutex send_audio_mutex_; | |
| // DTMF. | |
| bool dtmf_event_is_on_ = false; | |
| bool dtmf_event_first_packet_sent_ = false; | |
| int8_t dtmf_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_mutex_) = 8000; | |
| uint32_t dtmf_timestamp_ = 0; | |
| uint32_t dtmf_length_samples_ = 0; | |
| int64_t dtmf_time_last_sent_ = 0; | |
| uint32_t dtmf_timestamp_last_sent_ = 0; | |
| DtmfQueue::Event dtmf_current_event_; | |
| DtmfQueue dtmf_queue_; | |
| // VAD detection, used for marker bit. | |
| bool inband_vad_active_ RTC_GUARDED_BY(send_audio_mutex_) = false; | |
| int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; | |
| OneTimeEvent first_packet_sent_; | |
| std::optional<int> encoder_rtp_timestamp_frequency_ | |
| RTC_GUARDED_BY(send_audio_mutex_); | |
| AbsoluteCaptureTimeSender absolute_capture_time_sender_ | |
| RTC_GUARDED_BY(send_audio_mutex_); | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |