| /dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_coding/test/ |
| H A D | EncodeDecodeTest.cc | 33 TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency) in TestPacketization()
53 RTPStream* rtpStream, in Setup()
105 RTPStream* rtpStream, in Setup()
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| H A D | PacketLossTest.cc | 30 RTPStream* rtpStream, in Setup()
91 RTPStream* rtpStream, in Setup()
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| /dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_coding/test/ |
| H A D | EncodeDecodeTest.cc | 33 TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency) in TestPacketization()
53 RTPStream* rtpStream, in Setup()
105 RTPStream* rtpStream, in Setup()
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| H A D | PacketLossTest.cc | 30 RTPStream* rtpStream, in Setup()
91 RTPStream* rtpStream, in Setup()
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| /dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_coding/test/ |
| H A D | EncodeDecodeTest.cc | 33 TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency) in TestPacketization()
53 RTPStream* rtpStream, in Setup()
105 RTPStream* rtpStream, in Setup()
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| H A D | PacketLossTest.cc | 30 RTPStream* rtpStream, in Setup()
91 RTPStream* rtpStream, in Setup()
|
| /dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_coding/test/ |
| H A D | EncodeDecodeTest.cc | 33 TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency) in TestPacketization()
53 RTPStream* rtpStream, in Setup()
105 RTPStream* rtpStream, in Setup()
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| H A D | PacketLossTest.cc | 30 RTPStream* rtpStream, in Setup()
91 RTPStream* rtpStream, in Setup()
|
| /dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
| H A D | PacketLossTest.cc | 29 RTPStream *rtpStream, in Setup()
90 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
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| H A D | EncodeDecodeTest.cc | 27 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
53 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
124 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
| H A D | PacketLossTest.cc | 29 RTPStream *rtpStream, in Setup()
90 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
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| H A D | EncodeDecodeTest.cc | 29 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
55 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
126 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| /dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ |
| H A D | PacketLossTest.cc | 28 RTPStream *rtpStream, in Setup()
89 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| H A D | EncodeDecodeTest.cc | 28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
54 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
121 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| /dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
| H A D | PacketLossTest.cc | 29 RTPStream *rtpStream, in Setup()
90 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
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| H A D | EncodeDecodeTest.cc | 27 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
53 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
124 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| /dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
| H A D | PacketLossTest.cc | 29 RTPStream *rtpStream, in Setup()
90 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| H A D | EncodeDecodeTest.cc | 27 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
53 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
124 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| /dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/main/test/ |
| H A D | PacketLossTest.cc | 28 RTPStream *rtpStream, in Setup()
89 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
|
| H A D | EncodeDecodeTest.cc | 28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) in TestPacketization()
54 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
121 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, in Setup()
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| /dports/net/opal/opal-3.10.10/src/opal/ |
| H A D | rtpep.cxx | 84 const OpalRTPMediaStream * rtpStream = dynamic_cast<const OpalRTPMediaStream *>(&stream); in GetRTPFromStream() local
|