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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
| 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
| 13 | 13 |
| 14 #include "webrtc/base/criticalsection.h" | 14 #include "webrtc/base/criticalsection.h" |
| 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| 17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| 18 #include "webrtc/typedefs.h" | 18 #include "webrtc/typedefs.h" |
| 19 | 19 |
| 20 namespace webrtc { | 20 namespace webrtc { |
| 21 | 21 |
| 22 struct CodecInst; |
| 23 |
| 22 class TelephoneEventHandler; | 24 class TelephoneEventHandler; |
| 23 | 25 |
| 24 // This strategy deals with media-specific RTP packet processing. | 26 // This strategy deals with media-specific RTP packet processing. |
| 25 // This class is not thread-safe and must be protected by its caller. | 27 // This class is not thread-safe and must be protected by its caller. |
| 26 class RTPReceiverStrategy { | 28 class RTPReceiverStrategy { |
| 27 public: | 29 public: |
| 28 static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); | 30 static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); |
| 29 static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback); | 31 static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback); |
| 30 | 32 |
| 31 virtual ~RTPReceiverStrategy() {} | 33 virtual ~RTPReceiverStrategy() {} |
| (...skipping 15 matching lines...) Expand all Loading... |
| 47 virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; | 49 virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; |
| 48 | 50 |
| 49 // Computes the current dead-or-alive state. | 51 // Computes the current dead-or-alive state. |
| 50 virtual RTPAliveType ProcessDeadOrAlive( | 52 virtual RTPAliveType ProcessDeadOrAlive( |
| 51 uint16_t last_payload_length) const = 0; | 53 uint16_t last_payload_length) const = 0; |
| 52 | 54 |
| 53 // Returns true if we should report CSRC changes for this payload type. | 55 // Returns true if we should report CSRC changes for this payload type. |
| 54 // TODO(phoglund): should move out of here along with other payload stuff. | 56 // TODO(phoglund): should move out of here along with other payload stuff. |
| 55 virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0; | 57 virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0; |
| 56 | 58 |
| 57 // Notifies the strategy that we have created a new non-RED payload type in | 59 // Notifies the strategy that we have created a new non-RED audio payload type |
| 58 // the payload registry. | 60 // in the payload registry. |
| 59 virtual int32_t OnNewPayloadTypeCreated( | 61 virtual int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) = 0; |
| 60 const char payloadName[RTP_PAYLOAD_NAME_SIZE], | |
| 61 int8_t payloadType, | |
| 62 uint32_t frequency) = 0; | |
| 63 | 62 |
| 64 // Invokes the OnInitializeDecoder callback in a media-specific way. | 63 // Invokes the OnInitializeDecoder callback in a media-specific way. |
| 65 virtual int32_t InvokeOnInitializeDecoder( | 64 virtual int32_t InvokeOnInitializeDecoder( |
| 66 RtpFeedback* callback, | 65 RtpFeedback* callback, |
| 67 int8_t payload_type, | 66 int8_t payload_type, |
| 68 const char payload_name[RTP_PAYLOAD_NAME_SIZE], | 67 const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| 69 const PayloadUnion& specific_payload) const = 0; | 68 const PayloadUnion& specific_payload) const = 0; |
| 70 | 69 |
| 71 // Checks if the payload type has changed, and returns whether we should | 70 // Checks if the payload type has changed, and returns whether we should |
| 72 // reset statistics and/or discard this packet. | 71 // reset statistics and/or discard this packet. |
| (...skipping 18 matching lines...) Expand all Loading... |
| 91 // packet. | 90 // packet. |
| 92 explicit RTPReceiverStrategy(RtpData* data_callback); | 91 explicit RTPReceiverStrategy(RtpData* data_callback); |
| 93 | 92 |
| 94 rtc::CriticalSection crit_sect_; | 93 rtc::CriticalSection crit_sect_; |
| 95 PayloadUnion last_payload_; | 94 PayloadUnion last_payload_; |
| 96 RtpData* data_callback_; | 95 RtpData* data_callback_; |
| 97 }; | 96 }; |
| 98 } // namespace webrtc | 97 } // namespace webrtc |
| 99 | 98 |
| 100 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 99 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
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