blob: b8dec6c5e5286ac2fb75140ea997b326bd966ede [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/base/criticalsection.h"
#include "webrtc/base/onetimeevent.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/typedefs.h"
namespace webrtc {
class RTPSenderAudio : public DTMFqueue {
RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
virtual ~RTPSenderAudio();
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);
// set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG)
int32_t SetAudioPacketSize(uint16_t packet_size_samples);
// 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);
int AudioFrequency() const;
bool SendTelephoneEventPacket(
bool ended,
int8_t dtmf_payload_type,
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_;
RTPSender* const rtp_sender_;
rtc::CriticalSection send_audio_critsect_;
uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_);
// DTMF.
bool dtmf_event_is_on_;
bool dtmf_event_first_packet_sent_;
int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_);
uint32_t dtmf_timestamp_;
uint8_t dtmf_key_;
uint32_t dtmf_length_samples_;
uint8_t dtmf_level_;
int64_t dtmf_time_last_sent_;
uint32_t dtmf_timestamp_last_sent_;
// VAD detection, used for marker bit.
bool inband_vad_active_ GUARDED_BY(send_audio_critsect_);
int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_);
int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_);
int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_);
int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_);
int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_);
// Audio level indication.
// (
uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
OneTimeEvent first_packet_sent_;
} // namespace webrtc