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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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#include <set>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
// Handles audio RTP packets. This class is thread-safe.
class RTPReceiverAudio : public RTPReceiverStrategy,
public TelephoneEventHandler {
public:
RTPReceiverAudio(RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback);
virtual ~RTPReceiverAudio() {}
// The following three methods implement the TelephoneEventHandler interface.
// Forward DTMFs to decoder for playout.
void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
// Is forwarding of outband telephone events turned on/off?
bool TelephoneEventForwardToDecoder() const;
// Is TelephoneEvent configured with payload type payload_type
bool TelephoneEventPayloadType(const int8_t payload_type) const;
TelephoneEventHandler* GetTelephoneEventHandler() {
return this;
}
// Returns true if CNG is configured with payload type payload_type. If so,
// the frequency and cng_payload_type_has_changed are filled in.
bool CNGPayloadType(const int8_t payload_type,
uint32_t* frequency,
bool* cng_payload_type_has_changed);
int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* packet,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) override;
int GetPayloadTypeFrequency() const override;
RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
int32_t OnNewPayloadTypeCreated(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_type,
uint32_t frequency) override;
int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const override;
// We do not allow codecs to have multiple payload types for audio, so we
// need to override the default behavior (which is to do nothing).
void PossiblyRemoveExistingPayloadType(
RtpUtility::PayloadTypeMap* payload_type_map,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
size_t payload_name_length,
uint32_t frequency,
uint8_t channels,
uint32_t rate) const;
// 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;
int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
private:
int32_t ParseAudioCodecSpecific(
WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
size_t payload_length,
const AudioPayload& audio_specific,
bool is_red);
uint32_t last_received_frequency_;
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_;
int8_t cng_payload_type_;
// G722 is special since it use the wrong number of RTP samples in timestamp
// VS. number of samples in the frame
int8_t g722_payload_type_;
bool last_received_g722_;
uint8_t num_energy_;
uint8_t current_remote_energy_[kRtpCsrcSize];
RtpAudioFeedback* cb_audio_feedback_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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