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

Issue 1172163004: Reformat existing code. There should be no functional effects. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 6 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 43 matching lines...) Expand 10 before | Expand all | Expand 10 after
54 const RTPHeader* packet_header, 54 const RTPHeader* packet_header,
55 Modes prev_mode, 55 Modes prev_mode,
56 bool play_dtmf, 56 bool play_dtmf,
57 bool* reset_decoder) override; 57 bool* reset_decoder) override;
58 58
59 // Returns the operation to do given that the expected packet is not 59 // Returns the operation to do given that the expected packet is not
60 // available, but a packet further into the future is at hand. 60 // available, but a packet further into the future is at hand.
61 virtual Operations FuturePacketAvailable( 61 virtual Operations FuturePacketAvailable(
62 const SyncBuffer& sync_buffer, 62 const SyncBuffer& sync_buffer,
63 const Expand& expand, 63 const Expand& expand,
64 int decoder_frame_length, Modes prev_mode, 64 int decoder_frame_length,
65 Modes prev_mode,
65 uint32_t target_timestamp, 66 uint32_t target_timestamp,
66 uint32_t available_timestamp, 67 uint32_t available_timestamp,
67 bool play_dtmf); 68 bool play_dtmf);
68 69
69 // Returns the operation to do given that the expected packet is available. 70 // Returns the operation to do given that the expected packet is available.
70 virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf); 71 virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
71 72
72 // Returns the operation given that no packets are available (except maybe 73 // Returns the operation given that no packets are available (except maybe
73 // a DTMF event, flagged by setting |play_dtmf| true). 74 // a DTMF event, flagged by setting |play_dtmf| true).
74 virtual Operations NoPacket(bool play_dtmf); 75 virtual Operations NoPacket(bool play_dtmf);
(...skipping 21 matching lines...) Expand all
96 bool PacketTooEarly(uint32_t timestamp_leap) const; 97 bool PacketTooEarly(uint32_t timestamp_leap) const;
97 98
98 // Checks if num_consecutive_expands_ >= kMaxWaitForPacket. 99 // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
99 bool MaxWaitForPacket() const; 100 bool MaxWaitForPacket() const;
100 101
101 DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal); 102 DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
102 }; 103 };
103 104
104 } // namespace webrtc 105 } // namespace webrtc
105 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_ 106 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
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