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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_RECEIVER_AUDIO_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#include <set>
#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/onetimeevent.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
// Handles audio RTP packets. This class is thread-safe.
class RTPReceiverAudio : public RTPReceiverStrategy,
public TelephoneEventHandler {
public:
explicit RTPReceiverAudio(RtpData* data_callback);
~RTPReceiverAudio() override;
// The following three methods implement the TelephoneEventHandler interface.
// Forward DTMFs to decoder for playout.
void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
// Is forwarding of outband telephone events turned on/off?
bool TelephoneEventForwardToDecoder() const override;
// Is TelephoneEvent configured with |payload_type|.
bool TelephoneEventPayloadType(const int8_t payload_type) const override;
TelephoneEventHandler* GetTelephoneEventHandler() override;
// Returns true if CNG is configured with |payload_type|.
bool CNGPayloadType(const int8_t payload_type);
int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
const uint8_t* packet,
size_t payload_length,
int64_t timestamp_ms) override;
RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
int32_t OnNewPayloadTypeCreated(int payload_type,
const SdpAudioFormat& audio_format) override;
// We need to look out for special payload types here and sometimes reset
// statistics. In addition we sometimes need to tweak the frequency.
void CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
bool* should_discard_changes) override;
private:
int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
size_t payload_length,
const AudioPayload& audio_specific);
bool telephone_event_forward_to_decoder_;
int8_t telephone_event_payload_type_;
std::set<uint8_t> telephone_event_reported_;
int8_t cng_nb_payload_type_;
int8_t cng_wb_payload_type_;
int8_t cng_swb_payload_type_;
int8_t cng_fb_payload_type_;
ThreadUnsafeOneTimeEvent first_packet_received_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_