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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 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" | 11 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <sstream> | 14 #include <sstream> |
15 #include <stdio.h> | 15 #include <stdio.h> |
16 #include <stdlib.h> | 16 #include <stdlib.h> |
17 | 17 |
18 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
19 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" | 19 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" |
| 20 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
20 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 21 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
21 #include "webrtc/modules/audio_coding/test/utility.h" | 22 #include "webrtc/modules/audio_coding/test/utility.h" |
22 #include "webrtc/system_wrappers/include/trace.h" | 23 #include "webrtc/system_wrappers/include/trace.h" |
23 #include "webrtc/test/gtest.h" | 24 #include "webrtc/test/gtest.h" |
24 #include "webrtc/test/testsupport/fileutils.h" | 25 #include "webrtc/test/testsupport/fileutils.h" |
25 | 26 |
26 namespace webrtc { | 27 namespace webrtc { |
27 | 28 |
28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) | 29 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) |
29 : _rtpStream(rtpStream), | 30 : _rtpStream(rtpStream), |
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125 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, | 126 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, |
126 std::string out_file_name, size_t channels) { | 127 std::string out_file_name, size_t channels) { |
127 struct CodecInst recvCodec = CodecInst(); | 128 struct CodecInst recvCodec = CodecInst(); |
128 int noOfCodecs; | 129 int noOfCodecs; |
129 EXPECT_EQ(0, acm->InitializeReceiver()); | 130 EXPECT_EQ(0, acm->InitializeReceiver()); |
130 | 131 |
131 noOfCodecs = acm->NumberOfCodecs(); | 132 noOfCodecs = acm->NumberOfCodecs(); |
132 for (int i = 0; i < noOfCodecs; i++) { | 133 for (int i = 0; i < noOfCodecs; i++) { |
133 EXPECT_EQ(0, acm->Codec(i, &recvCodec)); | 134 EXPECT_EQ(0, acm->Codec(i, &recvCodec)); |
134 if (recvCodec.channels == channels) | 135 if (recvCodec.channels == channels) |
135 EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec)); | 136 EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype, |
| 137 CodecInstToSdp(recvCodec))); |
136 // Forces mono/stereo for Opus. | 138 // Forces mono/stereo for Opus. |
137 if (!strcmp(recvCodec.plname, "opus")) { | 139 if (!strcmp(recvCodec.plname, "opus")) { |
138 recvCodec.channels = channels; | 140 recvCodec.channels = channels; |
139 EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec)); | 141 EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype, |
| 142 CodecInstToSdp(recvCodec))); |
140 } | 143 } |
141 } | 144 } |
142 | 145 |
143 int playSampFreq; | 146 int playSampFreq; |
144 std::string file_name; | 147 std::string file_name; |
145 std::stringstream file_stream; | 148 std::stringstream file_stream; |
146 file_stream << webrtc::test::OutputPath() << out_file_name | 149 file_stream << webrtc::test::OutputPath() << out_file_name |
147 << static_cast<int>(codeId) << ".pcm"; | 150 << static_cast<int>(codeId) << ".pcm"; |
148 file_name = file_stream.str(); | 151 file_name = file_stream.str(); |
149 _rtpStream = rtpStream; | 152 _rtpStream = rtpStream; |
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350 if (acm->SendCodec()) { | 353 if (acm->SendCodec()) { |
351 _sender.Run(); | 354 _sender.Run(); |
352 } | 355 } |
353 _sender.Teardown(); | 356 _sender.Teardown(); |
354 rtpFile.Close(); | 357 rtpFile.Close(); |
355 | 358 |
356 return fileName; | 359 return fileName; |
357 } | 360 } |
358 | 361 |
359 } // namespace webrtc | 362 } // namespace webrtc |
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