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Side by Side Diff: webrtc/modules/audio_coding/neteq/decision_logic_normal.h

Issue 2411183003: Removed RTPHeader from NetEq's Packet struct. (Closed)
Patch Set: Fixed naming of payloadType and sequenceNumber. Updated comments. Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
(...skipping 26 matching lines...) Expand all
37 packet_buffer, 37 packet_buffer,
38 delay_manager, 38 delay_manager,
39 buffer_level_filter, 39 buffer_level_filter,
40 tick_timer) {} 40 tick_timer) {}
41 41
42 protected: 42 protected:
43 static const int kAllowMergeWithoutExpandMs = 20; // 20 ms. 43 static const int kAllowMergeWithoutExpandMs = 20; // 20 ms.
44 static const int kReinitAfterExpands = 100; 44 static const int kReinitAfterExpands = 100;
45 static const int kMaxWaitForPacket = 10; 45 static const int kMaxWaitForPacket = 10;
46 46
47 // Returns the operation that should be done next. |sync_buffer| and |expand|
48 // are provided for reference. |decoder_frame_length| is the number of samples
49 // obtained from the last decoded frame. If there is a packet available, the
50 // packet header should be supplied in |packet_header|; otherwise it should
51 // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
52 // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
53 // should be set to true. The output variable |reset_decoder| will be set to
54 // true if a reset is required; otherwise it is left unchanged (i.e., it can
55 // remain true if it was true before the call).
56 Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer, 47 Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
57 const Expand& expand, 48 const Expand& expand,
58 size_t decoder_frame_length, 49 size_t decoder_frame_length,
59 const RTPHeader* packet_header, 50 const Packet* next_packet,
60 Modes prev_mode, 51 Modes prev_mode,
61 bool play_dtmf, 52 bool play_dtmf,
62 bool* reset_decoder, 53 bool* reset_decoder,
63 size_t generated_noise_samples) override; 54 size_t generated_noise_samples) override;
64 55
65 // Returns the operation to do given that the expected packet is not 56 // Returns the operation to do given that the expected packet is not
66 // available, but a packet further into the future is at hand. 57 // available, but a packet further into the future is at hand.
67 virtual Operations FuturePacketAvailable( 58 virtual Operations FuturePacketAvailable(
68 const SyncBuffer& sync_buffer, 59 const SyncBuffer& sync_buffer,
69 const Expand& expand, 60 const Expand& expand,
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
108 bool PacketTooEarly(uint32_t timestamp_leap) const; 99 bool PacketTooEarly(uint32_t timestamp_leap) const;
109 100
110 // Checks if num_consecutive_expands_ >= kMaxWaitForPacket. 101 // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
111 bool MaxWaitForPacket() const; 102 bool MaxWaitForPacket() const;
112 103
113 RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal); 104 RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
114 }; 105 };
115 106
116 } // namespace webrtc 107 } // namespace webrtc
117 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ 108 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
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