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/*
* 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 <memory>
#include "absl/strings/string_view.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;
rtc::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`
absl::optional<Timestamp> capture_time;
// Audio level in dBov for
// header-extension-for-audio-level-indication.
// Valid range is [0,127]. Actual value is negative.
absl::optional<int> audio_level_dbov;
// Contributing sources list.
rtc::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_;
absl::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_
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