1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 13 14 #include <set> 15 16 #include "webrtc/base/scoped_ptr.h" 17 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 21 #include "webrtc/typedefs.h" 22 23 namespace webrtc { 24 25 class CriticalSectionWrapper; 26 27 // Handles audio RTP packets. This class is thread-safe. 28 class RTPReceiverAudio : public RTPReceiverStrategy, 29 public TelephoneEventHandler { 30 public: 31 RTPReceiverAudio(RtpData* data_callback, 32 RtpAudioFeedback* incoming_messages_callback); 33 virtual ~RTPReceiverAudio() {} 34 35 // The following three methods implement the TelephoneEventHandler interface. 36 // Forward DTMFs to decoder for playout. 37 void SetTelephoneEventForwardToDecoder(bool forward_to_decoder); 38 39 // Is forwarding of outband telephone events turned on/off? 40 bool TelephoneEventForwardToDecoder() const; 41 42 // Is TelephoneEvent configured with payload type payload_type 43 bool TelephoneEventPayloadType(const int8_t payload_type) const; 44 45 TelephoneEventHandler* GetTelephoneEventHandler() { return this; } 46 47 // Returns true if CNG is configured with payload type payload_type. If so, 48 // the frequency and cng_payload_type_has_changed are filled in. 49 bool CNGPayloadType(const int8_t payload_type, 50 uint32_t* frequency, 51 bool* cng_payload_type_has_changed); 52 53 int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, 54 const PayloadUnion& specific_payload, 55 bool is_red, 56 const uint8_t* packet, 57 size_t payload_length, 58 int64_t timestamp_ms, 59 bool is_first_packet) override; 60 61 int GetPayloadTypeFrequency() const override; 62 63 RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; 64 65 bool ShouldReportCsrcChanges(uint8_t payload_type) const override; 66 67 int32_t OnNewPayloadTypeCreated( 68 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 69 int8_t payload_type, 70 uint32_t frequency) override; 71 72 int32_t InvokeOnInitializeDecoder( 73 RtpFeedback* callback, 74 int8_t payload_type, 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 76 const PayloadUnion& specific_payload) const override; 77 78 // We do not allow codecs to have multiple payload types for audio, so we 79 // need to override the default behavior (which is to do nothing). 80 void PossiblyRemoveExistingPayloadType( 81 RtpUtility::PayloadTypeMap* payload_type_map, 82 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 83 size_t payload_name_length, 84 uint32_t frequency, 85 uint8_t channels, 86 uint32_t rate) const; 87 88 // We need to look out for special payload types here and sometimes reset 89 // statistics. In addition we sometimes need to tweak the frequency. 90 void CheckPayloadChanged(int8_t payload_type, 91 PayloadUnion* specific_payload, 92 bool* should_discard_changes) override; 93 94 int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override; 95 96 private: 97 int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header, 98 const uint8_t* payload_data, 99 size_t payload_length, 100 const AudioPayload& audio_specific, 101 bool is_red); 102 103 uint32_t last_received_frequency_; 104 105 bool telephone_event_forward_to_decoder_; 106 int8_t telephone_event_payload_type_; 107 std::set<uint8_t> telephone_event_reported_; 108 109 int8_t cng_nb_payload_type_; 110 int8_t cng_wb_payload_type_; 111 int8_t cng_swb_payload_type_; 112 int8_t cng_fb_payload_type_; 113 int8_t cng_payload_type_; 114 115 // G722 is special since it use the wrong number of RTP samples in timestamp 116 // VS. number of samples in the frame 117 int8_t g722_payload_type_; 118 bool last_received_g722_; 119 120 uint8_t num_energy_; 121 uint8_t current_remote_energy_[kRtpCsrcSize]; 122 123 RtpAudioFeedback* cb_audio_feedback_; 124 }; 125 } // namespace webrtc 126 127 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 128
