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Side by Side Diff: webrtc/modules/audio_coding/include/audio_coding_module.h

Issue 1484343003: NetEq: Add codec name and RTP timestamp rate to DecoderInfo (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years ago
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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_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
13 13
14 #include <string>
14 #include <vector> 15 #include <vector>
15 16
16 #include "webrtc/base/optional.h" 17 #include "webrtc/base/optional.h"
17 #include "webrtc/common_types.h" 18 #include "webrtc/common_types.h"
18 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" 19 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
19 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 20 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
20 #include "webrtc/modules/include/module.h" 21 #include "webrtc/modules/include/module.h"
21 #include "webrtc/system_wrappers/include/clock.h" 22 #include "webrtc/system_wrappers/include/clock.h"
22 #include "webrtc/typedefs.h" 23 #include "webrtc/typedefs.h"
23 24
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467 // 468 //
468 // Return value: 469 // Return value:
469 // -1 if failed to register the codec 470 // -1 if failed to register the codec
470 // 0 if the codec registered successfully. 471 // 0 if the codec registered successfully.
471 // 472 //
472 virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0; 473 virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
473 474
474 virtual int RegisterExternalReceiveCodec(int rtp_payload_type, 475 virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
475 AudioDecoder* external_decoder, 476 AudioDecoder* external_decoder,
476 int sample_rate_hz, 477 int sample_rate_hz,
477 int num_channels) = 0; 478 int num_channels,
479 const std::string& name) = 0;
kwiberg-webrtc 2015/12/02 09:50:42 You don't explain what the name is used for. (This
hlundin-webrtc 2015/12/02 16:19:24 Done.
478 480
479 /////////////////////////////////////////////////////////////////////////// 481 ///////////////////////////////////////////////////////////////////////////
480 // int32_t UnregisterReceiveCodec() 482 // int32_t UnregisterReceiveCodec()
481 // Unregister the codec currently registered with a specific payload type 483 // Unregister the codec currently registered with a specific payload type
482 // from the list of possible receive codecs. 484 // from the list of possible receive codecs.
483 // 485 //
484 // Input: 486 // Input:
485 // -payload_type : The number representing the payload type to 487 // -payload_type : The number representing the payload type to
486 // unregister. 488 // unregister.
487 // 489 //
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732 virtual std::vector<uint16_t> GetNackList( 734 virtual std::vector<uint16_t> GetNackList(
733 int64_t round_trip_time_ms) const = 0; 735 int64_t round_trip_time_ms) const = 0;
734 736
735 virtual void GetDecodingCallStatistics( 737 virtual void GetDecodingCallStatistics(
736 AudioDecodingCallStats* call_stats) const = 0; 738 AudioDecodingCallStats* call_stats) const = 0;
737 }; 739 };
738 740
739 } // namespace webrtc 741 } // namespace webrtc
740 742
741 #endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ 743 #endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
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