/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
D | rtp_payload_registry.cc | 39 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in RegisterReceivePayload() argument 111 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); in RegisterReceivePayload() 117 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); in RegisterReceivePayload() 147 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType() argument 180 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in ReceivePayloadType() argument 384 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in CreatePayloadType() argument 390 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CreatePayloadType() 391 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in CreatePayloadType() 422 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in CreatePayloadType() argument 441 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CreatePayloadType() [all …]
|
D | 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],
|
D | rtp_receiver_impl.cc | 99 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in RegisterReceivePayload() argument 268 char payload_name[RTP_PAYLOAD_NAME_SIZE]; in CheckSSRCChanged() 300 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CheckSSRCChanged() 301 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); in CheckSSRCChanged() 344 char payload_name[RTP_PAYLOAD_NAME_SIZE]; in CheckPayloadChanged() 391 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in CheckPayloadChanged() 392 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); in CheckPayloadChanged()
|
D | rtp_receiver_video.h | 47 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 55 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver_strategy.h | 66 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_receiver_video.cc | 44 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in OnNewPayloadTypeCreated() argument 97 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in InvokeOnInitializeDecoder() argument
|
D | rtp_sender_video.cc | 77 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in RegisterVideoPayload() argument 94 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in RegisterVideoPayload() 95 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in RegisterVideoPayload()
|
D | rtp_sender_audio.cc | 87 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in RegisterAudioPayload() argument 125 payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; in RegisterAudioPayload() 126 strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); in RegisterAudioPayload()
|
D | rtp_receiver_audio.cc | 157 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in OnNewPayloadTypeCreated() argument 272 const char payload_name[RTP_PAYLOAD_NAME_SIZE], in InvokeOnInitializeDecoder() argument
|
D | rtp_sender_unittest.cc | 739 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 826 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 881 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 965 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; in TEST_F() 1014 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; in TEST_F() 1043 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; in TEST_F() 1083 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event"; in TEST_F()
|
D | rtp_utility.h | 40 char name[RTP_PAYLOAD_NAME_SIZE];
|
D | rtp_receiver_impl.h | 38 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 | 40 int32_t RegisterVideoPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
D | rtp_sender.h | 97 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
D | 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],
|
D | rtp_rtcp_defines.h | 262 const char payloadName[RTP_PAYLOAD_NAME_SIZE], 333 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in OnInitializeDecoder() argument
|
D | rtp_receiver.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
D | before_streaming_fixture.cc | 70 _snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE - 1, "PCMU"); in SetUpLocalPlayback() 72 snprintf(codec.plname, RTP_PAYLOAD_NAME_SIZE, "PCMU"); in SetUpLocalPlayback()
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/mock/ |
D | mock_rtp_payload_strategy.h | 34 RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
D | TestSenderReceiver.h | 77 const int8_t payloadName[RTP_PAYLOAD_NAME_SIZE], in OnInitializeDecoder() argument
|
/external/chromium_org/third_party/webrtc/ |
D | common_types.h | 40 #define RTP_PAYLOAD_NAME_SIZE 32 macro 288 char plname[RTP_PAYLOAD_NAME_SIZE];
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api_audio.cc | 67 const char payloadName[RTP_PAYLOAD_NAME_SIZE], in OnInitializeDecoder() argument
|
/external/chromium_org/third_party/webrtc/video_engine/ |
D | vie_channel.h | 240 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
/external/chromium_org/third_party/webrtc/voice_engine/ |
D | channel.h | 378 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|