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Side by Side Diff: webrtc/modules/audio_coding/main/interface/audio_coding_module.h

Issue 1312493004: Add support for external decoders in ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@isac-lock-2
Patch Set: Created 5 years, 3 months 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
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22 #include "webrtc/typedefs.h" 22 #include "webrtc/typedefs.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
25 25
26 // forward declarations 26 // forward declarations
27 struct CodecInst; 27 struct CodecInst;
28 struct WebRtcRTPHeader; 28 struct WebRtcRTPHeader;
29 class AudioFrame; 29 class AudioFrame;
30 class RTPFragmentationHeader; 30 class RTPFragmentationHeader;
31 class AudioEncoderMutable; 31 class AudioEncoderMutable;
32 class AudioDecoder;
32 33
33 #define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz 34 #define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
34 35
35 // Callback class used for sending data ready to be packetized 36 // Callback class used for sending data ready to be packetized
36 class AudioPacketizationCallback { 37 class AudioPacketizationCallback {
37 public: 38 public:
38 virtual ~AudioPacketizationCallback() {} 39 virtual ~AudioPacketizationCallback() {}
39 40
40 virtual int32_t SendData(FrameType frame_type, 41 virtual int32_t SendData(FrameType frame_type,
41 uint8_t payload_type, 42 uint8_t payload_type,
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569 // 570 //
570 // Input: 571 // Input:
571 // -receive_codec : parameters of the codec to be registered, c.f. 572 // -receive_codec : parameters of the codec to be registered, c.f.
572 // common_types.h for the definition of 573 // common_types.h for the definition of
573 // CodecInst. 574 // CodecInst.
574 // 575 //
575 // Return value: 576 // Return value:
576 // -1 if failed to register the codec 577 // -1 if failed to register the codec
577 // 0 if the codec registered successfully. 578 // 0 if the codec registered successfully.
578 // 579 //
579 virtual int32_t RegisterReceiveCodec( 580 virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
580 const CodecInst& receive_codec) = 0; 581
582 virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
583 AudioDecoder* external_decoder,
584 int sample_rate_hz,
585 int num_channels) = 0;
581 586
582 /////////////////////////////////////////////////////////////////////////// 587 ///////////////////////////////////////////////////////////////////////////
583 // int32_t UnregisterReceiveCodec() 588 // int32_t UnregisterReceiveCodec()
584 // Unregister the codec currently registered with a specific payload type 589 // Unregister the codec currently registered with a specific payload type
585 // from the list of possible receive codecs. 590 // from the list of possible receive codecs.
586 // 591 //
587 // Input: 592 // Input:
588 // -payload_type : The number representing the payload type to 593 // -payload_type : The number representing the payload type to
589 // unregister. 594 // unregister.
590 // 595 //
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1162 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0; 1167 virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
1163 1168
1164 // Returns the timing statistics for calls to Get10MsAudio. 1169 // Returns the timing statistics for calls to Get10MsAudio.
1165 virtual void GetDecodingCallStatistics( 1170 virtual void GetDecodingCallStatistics(
1166 AudioDecodingCallStats* call_stats) const = 0; 1171 AudioDecodingCallStats* call_stats) const = 0;
1167 }; 1172 };
1168 1173
1169 } // namespace webrtc 1174 } // namespace webrtc
1170 1175
1171 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_ 1176 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
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