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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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274 void EncodeDecodeTest::Perform() { | 274 void EncodeDecodeTest::Perform() { |
275 int numCodecs = 1; | 275 int numCodecs = 1; |
276 int codePars[3]; // Frequency, packet size, rate. | 276 int codePars[3]; // Frequency, packet size, rate. |
277 int numPars[52]; // Number of codec parameters sets (freq, pacsize, rate) | 277 int numPars[52]; // Number of codec parameters sets (freq, pacsize, rate) |
278 // to test, for a given codec. | 278 // to test, for a given codec. |
279 | 279 |
280 codePars[0] = 0; | 280 codePars[0] = 0; |
281 codePars[1] = 0; | 281 codePars[1] = 0; |
282 codePars[2] = 0; | 282 codePars[2] = 0; |
283 | 283 |
284 std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create()); | 284 std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0)); |
285 struct CodecInst sendCodecTmp; | 285 struct CodecInst sendCodecTmp; |
286 numCodecs = acm->NumberOfCodecs(); | 286 numCodecs = acm->NumberOfCodecs(); |
287 | 287 |
288 if (_testMode != 2) { | 288 if (_testMode != 2) { |
289 for (int n = 0; n < numCodecs; n++) { | 289 for (int n = 0; n < numCodecs; n++) { |
290 EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp)); | 290 EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp)); |
291 if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) { | 291 if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) { |
292 numPars[n] = 0; | 292 numPars[n] = 0; |
293 } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) { | 293 } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) { |
294 numPars[n] = 0; | 294 numPars[n] = 0; |
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330 // End tracing. | 330 // End tracing. |
331 if (_testMode == 1) { | 331 if (_testMode == 1) { |
332 Trace::ReturnTrace(); | 332 Trace::ReturnTrace(); |
333 } | 333 } |
334 } | 334 } |
335 | 335 |
336 std::string EncodeDecodeTest::EncodeToFile(int fileType, | 336 std::string EncodeDecodeTest::EncodeToFile(int fileType, |
337 int codeId, | 337 int codeId, |
338 int* codePars, | 338 int* codePars, |
339 int testMode) { | 339 int testMode) { |
340 std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create()); | 340 std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1)); |
341 RTPFile rtpFile; | 341 RTPFile rtpFile; |
342 std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), | 342 std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), |
343 "encode_decode_rtp"); | 343 "encode_decode_rtp"); |
344 rtpFile.Open(fileName.c_str(), "wb+"); | 344 rtpFile.Open(fileName.c_str(), "wb+"); |
345 rtpFile.WriteHeader(); | 345 rtpFile.WriteHeader(); |
346 | 346 |
347 // Store for auto_test and logging. | 347 // Store for auto_test and logging. |
348 _sender.testMode = testMode; | 348 _sender.testMode = testMode; |
349 _sender.codeId = codeId; | 349 _sender.codeId = codeId; |
350 | 350 |
351 _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1); | 351 _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1); |
352 if (acm->SendCodec()) { | 352 if (acm->SendCodec()) { |
353 _sender.Run(); | 353 _sender.Run(); |
354 } | 354 } |
355 _sender.Teardown(); | 355 _sender.Teardown(); |
356 rtpFile.Close(); | 356 rtpFile.Close(); |
357 | 357 |
358 return fileName; | 358 return fileName; |
359 } | 359 } |
360 | 360 |
361 } // namespace webrtc | 361 } // namespace webrtc |
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