/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtp_payload_registry.cc | 40 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in RegisterReceivePayload() argument 139 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType() argument 172 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in ReceivePayloadType() argument 407 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in CreatePayloadType() argument 413 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CreatePayloadType() 414 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in CreatePayloadType() 445 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in CreatePayloadType() argument 468 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CreatePayloadType() 469 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in CreatePayloadType()
|
D | rtp_receiver_impl.cc | 96 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in RegisterReceivePayload() argument 254 char payload_name[RTP_PAYLOAD_NAME_SIZE]; in CheckSSRCChanged() 283 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CheckSSRCChanged() 284 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); in CheckSSRCChanged() 326 char payload_name[RTP_PAYLOAD_NAME_SIZE]; in CheckPayloadChanged() 372 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CheckPayloadChanged() 373 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); in CheckPayloadChanged()
|
D | rtp_receiver_audio.h | 68 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 82 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver_video.h | 46 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 53 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver_strategy.h | 66 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 74 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver_video.cc | 46 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in OnNewPayloadTypeCreated() argument 115 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in InvokeOnInitializeDecoder() argument
|
D | rtp_sender_audio.cc | 66 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in RegisterAudioPayload() argument 104 (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0'; in RegisterAudioPayload() 105 strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in RegisterAudioPayload()
|
D | rtp_sender_video.cc | 74 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in CreateVideoPayload() argument 90 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CreateVideoPayload() 91 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in CreateVideoPayload()
|
D | rtp_utility.h | 35 char name[RTP_PAYLOAD_NAME_SIZE];
|
D | rtp_receiver_audio.cc | 155 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in OnNewPayloadTypeCreated() argument 265 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in InvokeOnInitializeDecoder() argument
|
D | rtp_receiver_impl.h | 36 int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_sender_audio.h | 29 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_sender_video.h | 43 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_sender_unittest.cc | 923 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 998 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 1057 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 1140 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 1207 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; in TEST_F() 1236 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; in TEST_F() 1277 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event"; in TEST_F()
|
D | rtp_sender.h | 117 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_sender.cc | 296 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in RegisterPayload() argument 314 payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) { in RegisterPayload()
|
/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
D | rtp_payload_registry.h | 39 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 72 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 173 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_rtcp_defines.h | 211 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 334 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in OnInitializeDecoder() argument
|
/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
D | before_streaming_fixture.cc | 74 _snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE - 1, "PCMU"); in SetUpLocalPlayback() 76 snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE, "PCMU"); in SetUpLocalPlayback()
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/mock/ |
D | mock_rtp_payload_strategy.h | 34 RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
/external/webrtc/webrtc/ |
D | common_types.h | 40 #define RTP_PAYLOAD_NAME_SIZE 32 macro 291 char plname[RTP_PAYLOAD_NAME_SIZE];
|
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api_audio.cc | 65 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in OnInitializeDecoder() argument
|
/external/webrtc/webrtc/video/ |
D | vie_channel.h | 191 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
/external/webrtc/webrtc/voice_engine/ |
D | channel.h | 382 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|