Line data    Source code

       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/onetimeevent.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             : // Handles audio RTP packets. This class is thread-safe.
      26             : class RTPReceiverAudio : public RTPReceiverStrategy,
      27             :                          public TelephoneEventHandler {
      28             :  public:
      29             :   explicit RTPReceiverAudio(RtpData* data_callback);
      30           0 :   virtual ~RTPReceiverAudio() {}
      31             : 
      32             :   // The following three methods implement the TelephoneEventHandler interface.
      33             :   // Forward DTMFs to decoder for playout.
      34             :   void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
      35             : 
      36             :   // Is forwarding of outband telephone events turned on/off?
      37             :   bool TelephoneEventForwardToDecoder() const override;
      38             : 
      39             :   // Is TelephoneEvent configured with |payload_type|.
      40             :   bool TelephoneEventPayloadType(const int8_t payload_type) const override;
      41             : 
      42           0 :   TelephoneEventHandler* GetTelephoneEventHandler() override { return this; }
      43             : 
      44             :   // Returns true if CNG is configured with |payload_type|.
      45             :   bool CNGPayloadType(const int8_t payload_type);
      46             : 
      47             :   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
      48             :                          const PayloadUnion& specific_payload,
      49             :                          bool is_red,
      50             :                          const uint8_t* packet,
      51             :                          size_t payload_length,
      52             :                          int64_t timestamp_ms,
      53             :                          bool is_first_packet) override;
      54             : 
      55             :   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
      56             : 
      57             :   bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
      58             : 
      59             :   int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
      60             : 
      61             :   int32_t InvokeOnInitializeDecoder(
      62             :       RtpFeedback* callback,
      63             :       int8_t payload_type,
      64             :       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
      65             :       const PayloadUnion& specific_payload) const override;
      66             : 
      67             :   // We need to look out for special payload types here and sometimes reset
      68             :   // statistics. In addition we sometimes need to tweak the frequency.
      69             :   void CheckPayloadChanged(int8_t payload_type,
      70             :                            PayloadUnion* specific_payload,
      71             :                            bool* should_discard_changes) override;
      72             : 
      73             :   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
      74             : 
      75             :  private:
      76             :   int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
      77             :                                   const uint8_t* payload_data,
      78             :                                   size_t payload_length,
      79             :                                   const AudioPayload& audio_specific,
      80             :                                   bool is_red);
      81             : 
      82             :   bool telephone_event_forward_to_decoder_;
      83             :   int8_t telephone_event_payload_type_;
      84             :   std::set<uint8_t> telephone_event_reported_;
      85             : 
      86             :   int8_t cng_nb_payload_type_;
      87             :   int8_t cng_wb_payload_type_;
      88             :   int8_t cng_swb_payload_type_;
      89             :   int8_t cng_fb_payload_type_;
      90             : 
      91             :   uint8_t num_energy_;
      92             :   uint8_t current_remote_energy_[kRtpCsrcSize];
      93             : 
      94             :   ThreadUnsafeOneTimeEvent first_packet_received_;
      95             : };
      96             : }  // namespace webrtc
      97             : 
      98             : #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_