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Side by Side Diff: webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
(...skipping 10 matching lines...) Expand all
21 21
22 class ExtensionVerifyTransport : public webrtc::Transport { 22 class ExtensionVerifyTransport : public webrtc::Transport {
23 public: 23 public:
24 ExtensionVerifyTransport() 24 ExtensionVerifyTransport()
25 : parser_(webrtc::RtpHeaderParser::Create()), 25 : parser_(webrtc::RtpHeaderParser::Create()),
26 received_packets_(0), 26 received_packets_(0),
27 bad_packets_(0), 27 bad_packets_(0),
28 audio_level_id_(-1), 28 audio_level_id_(-1),
29 absolute_sender_time_id_(-1) {} 29 absolute_sender_time_id_(-1) {}
30 30
31 int SendPacket(int channel, const void* data, size_t len) override { 31 int SendPacket(const void* data, size_t len) override {
32 webrtc::RTPHeader header; 32 webrtc::RTPHeader header;
33 if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) { 33 if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
34 bool ok = true; 34 bool ok = true;
35 if (audio_level_id_ >= 0 && 35 if (audio_level_id_ >= 0 &&
36 !header.extension.hasAudioLevel) { 36 !header.extension.hasAudioLevel) {
37 ok = false; 37 ok = false;
38 } 38 }
39 if (absolute_sender_time_id_ >= 0 && 39 if (absolute_sender_time_id_ >= 0 &&
40 !header.extension.hasAbsoluteSendTime) { 40 !header.extension.hasAbsoluteSendTime) {
41 ok = false; 41 ok = false;
42 } 42 }
43 if (!ok) { 43 if (!ok) {
44 // bad_packets_ count packets we expected to have an extension but 44 // bad_packets_ count packets we expected to have an extension but
45 // didn't have one. 45 // didn't have one.
46 ++bad_packets_; 46 ++bad_packets_;
47 } 47 }
48 } 48 }
49 // received_packets_ count all packets we receive. 49 // received_packets_ count all packets we receive.
50 ++received_packets_; 50 ++received_packets_;
51 return static_cast<int>(len); 51 return static_cast<int>(len);
52 } 52 }
53 53
54 int SendRTCPPacket(int channel, const void* data, size_t len) override { 54 int SendRTCPPacket(const void* data, size_t len) override {
55 return static_cast<int>(len); 55 return static_cast<int>(len);
56 } 56 }
57 57
58 void SetAudioLevelId(int id) { 58 void SetAudioLevelId(int id) {
59 audio_level_id_ = id; 59 audio_level_id_ = id;
60 parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, id); 60 parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, id);
61 } 61 }
62 62
63 void SetAbsoluteSenderTimeId(int id) { 63 void SetAbsoluteSenderTimeId(int id) {
64 absolute_sender_time_id_ = id; 64 absolute_sender_time_id_ = id;
(...skipping 79 matching lines...) Expand 10 before | Expand all | Expand 10 after
144 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 144 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
145 3)); 145 3));
146 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true, 146 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
147 9)); 147 9));
148 verifying_transport_.SetAbsoluteSenderTimeId(3); 148 verifying_transport_.SetAbsoluteSenderTimeId(3);
149 // Don't register audio level with header parser - unknown extensions should 149 // Don't register audio level with header parser - unknown extensions should
150 // be ignored when parsing. 150 // be ignored when parsing.
151 ResumePlaying(); 151 ResumePlaying();
152 EXPECT_TRUE(verifying_transport_.Wait()); 152 EXPECT_TRUE(verifying_transport_.Wait());
153 } 153 }
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