rtp_sender_audio.cc | Explore in Territory

/*
 *  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 "modules/rtp_rtcp/source/rtp_sender_audio.h"

#include <string.h>

#include <memory>
#include <utility>
#include <vector>

#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/ntp_time_util.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "system_wrappers/include/ntp_time.h"

namespace webrtc {

RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
    : … { … }

RTPSenderAudio::~RTPSenderAudio() { … }

int32_t RTPSenderAudio::RegisterAudioPayload(absl::string_view payload_name,
                                             const int8_t payload_type,
                                             const uint32_t frequency,
                                             const size_t channels,
                                             const uint32_t rate) { … }

bool RTPSenderAudio::MarkerBit(AudioFrameType frame_type, int8_t payload_type) { … }

bool RTPSenderAudio::SendAudio(const RtpAudioFrame& frame) { … }

// Send a TelephoneEvent tone using RFC 2833 (4733)
int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                           uint16_t time_ms,
                                           uint8_t level) { … }

bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                                              uint32_t dtmf_timestamp,
                                              uint16_t duration,
                                              bool marker_bit) { … }
}  // namespace webrtc
chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
