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

Issue 1406123011: Let AudioCodingModule::SendCodec return Maybe<CodecInst> (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: review fix Created 5 years, 1 month 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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332 std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), 332 std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
333 "encode_decode_rtp"); 333 "encode_decode_rtp");
334 rtpFile.Open(fileName.c_str(), "wb+"); 334 rtpFile.Open(fileName.c_str(), "wb+");
335 rtpFile.WriteHeader(); 335 rtpFile.WriteHeader();
336 336
337 // Store for auto_test and logging. 337 // Store for auto_test and logging.
338 _sender.testMode = testMode; 338 _sender.testMode = testMode;
339 _sender.codeId = codeId; 339 _sender.codeId = codeId;
340 340
341 _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1); 341 _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
342 struct CodecInst sendCodecInst; 342 if (acm->SendCodec()) {
343 if (acm->SendCodec(&sendCodecInst) >= 0) {
344 _sender.Run(); 343 _sender.Run();
345 } 344 }
346 _sender.Teardown(); 345 _sender.Teardown();
347 rtpFile.Close(); 346 rtpFile.Close();
348 347
349 return fileName; 348 return fileName;
350 } 349 }
351 350
352 } // namespace webrtc 351 } // namespace webrtc
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