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      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/modules/rtp_rtcp/interface/rtp_receiver.h"
     17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
     18 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
     19 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
     20 #include "webrtc/system_wrappers/interface/scoped_ptr.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(const int32_t id,
     32                    RtpData* data_callback,
     33                    RtpAudioFeedback* incoming_messages_callback);
     34   virtual ~RTPReceiverAudio() {}
     35 
     36   // The following three methods implement the TelephoneEventHandler interface.
     37   // Forward DTMFs to decoder for playout.
     38   void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
     39 
     40   // Is forwarding of outband telephone events turned on/off?
     41   bool TelephoneEventForwardToDecoder() const;
     42 
     43   // Is TelephoneEvent configured with payload type payload_type
     44   bool TelephoneEventPayloadType(const int8_t payload_type) const;
     45 
     46   TelephoneEventHandler* GetTelephoneEventHandler() {
     47     return this;
     48   }
     49 
     50   // Returns true if CNG is configured with payload type payload_type. If so,
     51   // the frequency and cng_payload_type_has_changed are filled in.
     52   bool CNGPayloadType(const int8_t payload_type,
     53                       uint32_t* frequency,
     54                       bool* cng_payload_type_has_changed);
     55 
     56   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
     57                          const PayloadUnion& specific_payload,
     58                          bool is_red,
     59                          const uint8_t* packet,
     60                          uint16_t packet_length,
     61                          int64_t timestamp_ms,
     62                          bool is_first_packet);
     63 
     64   int GetPayloadTypeFrequency() const OVERRIDE;
     65 
     66   virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
     67       OVERRIDE;
     68 
     69   virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
     70 
     71   virtual int32_t OnNewPayloadTypeCreated(
     72       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     73       int8_t payload_type,
     74       uint32_t frequency) OVERRIDE;
     75 
     76   virtual int32_t InvokeOnInitializeDecoder(
     77       RtpFeedback* callback,
     78       int32_t id,
     79       int8_t payload_type,
     80       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     81       const PayloadUnion& specific_payload) const OVERRIDE;
     82 
     83   // We do not allow codecs to have multiple payload types for audio, so we
     84   // need to override the default behavior (which is to do nothing).
     85   void PossiblyRemoveExistingPayloadType(
     86       ModuleRTPUtility::PayloadTypeMap* payload_type_map,
     87       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     88       size_t payload_name_length,
     89       uint32_t frequency,
     90       uint8_t channels,
     91       uint32_t rate) const;
     92 
     93   // We need to look out for special payload types here and sometimes reset
     94   // statistics. In addition we sometimes need to tweak the frequency.
     95   void CheckPayloadChanged(int8_t payload_type,
     96                            PayloadUnion* specific_payload,
     97                            bool* should_reset_statistics,
     98                            bool* should_discard_changes) OVERRIDE;
     99 
    100   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const OVERRIDE;
    101 
    102  private:
    103 
    104   int32_t ParseAudioCodecSpecific(
    105       WebRtcRTPHeader* rtp_header,
    106       const uint8_t* payload_data,
    107       uint16_t payload_length,
    108       const AudioPayload& audio_specific,
    109       bool is_red);
    110 
    111   int32_t id_;
    112 
    113   uint32_t last_received_frequency_;
    114 
    115   bool telephone_event_forward_to_decoder_;
    116   int8_t telephone_event_payload_type_;
    117   std::set<uint8_t> telephone_event_reported_;
    118 
    119   int8_t cng_nb_payload_type_;
    120   int8_t cng_wb_payload_type_;
    121   int8_t cng_swb_payload_type_;
    122   int8_t cng_fb_payload_type_;
    123   int8_t cng_payload_type_;
    124 
    125   // G722 is special since it use the wrong number of RTP samples in timestamp
    126   // VS. number of samples in the frame
    127   int8_t g722_payload_type_;
    128   bool last_received_g722_;
    129 
    130   uint8_t num_energy_;
    131   uint8_t current_remote_energy_[kRtpCsrcSize];
    132 
    133   RtpAudioFeedback* cb_audio_feedback_;
    134 };
    135 }  // namespace webrtc
    136 
    137 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
    138