blob: 3a070805995f1dde5edd874ef4b904cbd400748b [file] [log] [blame]
* 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.
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/onetimeevent.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RTPSenderAudio {
RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_type,
uint32_t frequency,
size_t channels,
uint32_t rate,
RtpUtility::Payload** payload);
bool SendAudio(FrameType frame_type,
int8_t payload_type,
uint32_t capture_timestamp,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation);
// Store the audio level in dBov for
// header-extension-for-audio-level-indication.
// Valid range is [0,100]. Actual value is negative.
int32_t SetAudioLevel(uint8_t level_dbov);
// Send a DTMF tone using RFC 2833 (4733)
int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
bool SendTelephoneEventPacket(
bool ended,
uint32_t dtmf_timestamp,
uint16_t duration,
bool marker_bit); // set on first packet in talk burst
bool MarkerBit(FrameType frame_type, int8_t payload_type);
Clock* const clock_ = nullptr;
RTPSender* const rtp_sender_ = nullptr;
rtc::CriticalSection send_audio_critsect_;
// DTMF.
bool dtmf_event_is_on_ = false;
bool dtmf_event_first_packet_sent_ = false;
int8_t dtmf_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_critsect_) = 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_critsect_) = false;
int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_critsect_) = -1;
// Audio level indication.
// (
uint8_t audio_level_dbov_ RTC_GUARDED_BY(send_audio_critsect_) = 0;
OneTimeEvent first_packet_sent_;
} // namespace webrtc