/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_receiver_video.h | 47 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 55 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_receiver_audio.h | 72 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 80 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 87 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_receiver_strategy.h | 66 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_payload_registry.cc | 39 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 111 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); 117 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); 147 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 180 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 384 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 390 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 391 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); 422 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 441 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0 [all...] |
rtp_receiver_video.cc | 44 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 97 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_utility.h | 40 char name[RTP_PAYLOAD_NAME_SIZE];
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rtp_receiver_impl.cc | 99 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 268 char payload_name[RTP_PAYLOAD_NAME_SIZE]; 300 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 301 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); 344 char payload_name[RTP_PAYLOAD_NAME_SIZE]; 391 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 392 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
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rtp_receiver_impl.h | 38 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_sender_audio.h | 29 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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rtp_sender_video.h | 40 int32_t RegisterVideoPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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rtp_sender_audio.cc | 87 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 125 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 126 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
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rtp_sender_video.cc | 77 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 94 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 95 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
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rtp_receiver_audio.cc | 157 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 272 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_sender_unittest.cc | 739 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; [all...] |
rtp_sender.h | 97 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
rtp_payload_registry.h | 37 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 59 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 70 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 142 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_receiver.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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rtp_rtcp_defines.h | 262 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 333 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/mock/ |
mock_rtp_payload_strategy.h | 34 RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
before_streaming_fixture.cc | 70 _snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE - 1, "PCMU"); 72 snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE, "PCMU");
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
TestSenderReceiver.h | 77 const int8_t payloadName[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/ |
common_types.h | 40 #define RTP_PAYLOAD_NAME_SIZE 32 288 char plname[RTP_PAYLOAD_NAME_SIZE];
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api_audio.cc | 67 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/video_engine/ |
vie_channel.h | 240 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/webrtc/voice_engine/ |
channel.h | 378 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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