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Side by Side Diff: webrtc/modules/audio_coding/main/test/TestAllCodecs.cc

Issue 1371043003: Unify FrameType and VideoFrameType. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 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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67 rtp_info.header.markerBit = false; 67 rtp_info.header.markerBit = false;
68 rtp_info.header.ssrc = 0; 68 rtp_info.header.ssrc = 0;
69 rtp_info.header.sequenceNumber = sequence_number_++; 69 rtp_info.header.sequenceNumber = sequence_number_++;
70 rtp_info.header.payloadType = payload_type; 70 rtp_info.header.payloadType = payload_type;
71 rtp_info.header.timestamp = timestamp; 71 rtp_info.header.timestamp = timestamp;
72 if (frame_type == kAudioFrameCN) { 72 if (frame_type == kAudioFrameCN) {
73 rtp_info.type.Audio.isCNG = true; 73 rtp_info.type.Audio.isCNG = true;
74 } else { 74 } else {
75 rtp_info.type.Audio.isCNG = false; 75 rtp_info.type.Audio.isCNG = false;
76 } 76 }
77 if (frame_type == kFrameEmpty) { 77 if (frame_type == kEmptyFrame) {
78 // Skip this frame. 78 // Skip this frame.
79 return 0; 79 return 0;
80 } 80 }
81 81
82 // Only run mono for all test cases. 82 // Only run mono for all test cases.
83 rtp_info.type.Audio.channel = 1; 83 rtp_info.type.Audio.channel = 1;
84 memcpy(payload_data_, payload_data, payload_size); 84 memcpy(payload_data_, payload_data, payload_size);
85 85
86 status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info); 86 status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
87 87
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477 477
478 void TestAllCodecs::DisplaySendReceiveCodec() { 478 void TestAllCodecs::DisplaySendReceiveCodec() {
479 CodecInst my_codec_param; 479 CodecInst my_codec_param;
480 acm_a_->SendCodec(&my_codec_param); 480 acm_a_->SendCodec(&my_codec_param);
481 printf("%s -> ", my_codec_param.plname); 481 printf("%s -> ", my_codec_param.plname);
482 acm_b_->ReceiveCodec(&my_codec_param); 482 acm_b_->ReceiveCodec(&my_codec_param);
483 printf("%s\n", my_codec_param.plname); 483 printf("%s\n", my_codec_param.plname);
484 } 484 }
485 485
486 } // namespace webrtc 486 } // namespace webrtc
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