/external/chromium_org/third_party/webrtc/video_engine/ |
vie_channel.h | 147 int SetRtxSendPayloadType(int payload_type); 150 void SetRtxReceivePayloadType(int payload_type); 235 const int8_t payload_type,
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vie_receiver.h | 50 void SetRtxPayloadType(int payload_type);
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vie_encoder.h | 126 uint8_t payload_type,
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vie_receiver.cc | 99 void ViEReceiver::SetRtxPayloadType(int payload_type) { 100 rtp_payload_registry_->SetRtxPayloadType(payload_type);
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vie_rtp_rtcp_impl.cc | 195 const uint8_t payload_type) { 197 << " payload_type: " << payload_type; local 204 if (vie_channel->SetRtxSendPayloadType(payload_type) != 0) { 226 const uint8_t payload_type) { 228 << " payload_type: " << payload_type; local 235 vie_channel->SetRtxReceivePayloadType(payload_type); [all...] |
vie_channel.cc | 870 int payload_type; local 871 rtp_rtcp_->RTXSendStatus(&mode, &ssrc, &payload_type); 879 int ViEChannel::SetRtxSendPayloadType(int payload_type) { 880 rtp_rtcp_->SetRtxSendPayloadType(payload_type); 883 (*it)->SetRtxSendPayloadType(payload_type); 905 void ViEChannel::SetRtxReceivePayloadType(int payload_type) { 906 vie_receiver_.SetRtxPayloadType(payload_type); [all...] |
/external/chromium_org/media/cast/audio_sender/ |
audio_sender_unittest.cc | 73 audio_config_.rtp_config.payload_type = 127;
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/external/chromium_org/media/cast/transport/ |
cast_transport_config.h | 48 int payload_type; member in struct:media::cast::transport::RtpConfig
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_codec_database.h | 339 // [payload_type] - payload type. 340 static bool ValidPayloadType(int payload_type);
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audio_coding_module_unittest.cc | 41 RtpUtility(int samples_per_packet, uint8_t payload_type) 42 : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} 78 uint8_t payload_type,
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audio_coding_module_impl.cc | 287 int payload_type, 301 fragmentation_.fragmentationPlType[fragmentation_index] = payload_type; 1684 uint8_t payload_type = static_cast<uint8_t>(codec.pltype); local [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_sender_video.h | 100 const int8_t payload_type,
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rtp_sender_video.cc | 334 const int8_t payload_type, 364 if (_rtpSender.BuildRTPheader(buffer, payload_type, size == 0,
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/external/ipsec-tools/src/racoon/ |
isakmp_var.h | 110 vchar_t *payload, int payload_type));
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
rtputils.cc | 156 return (GetRtpPayloadType(data, len, &(header->payload_type)) && 220 SetRtpPayloadType(data, len, header.payload_type) &&
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rtpdataengine.cc | 262 if (!FindCodecById(recv_codecs_, header.payload_type, &codec)) { 268 // << " because unknown payload id: " << header.payload_type; 351 header.payload_type = found_codec.id;
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/external/chromium_org/third_party/webrtc/video_engine/include/ |
vie_rtp_rtcp.h | 135 const uint8_t payload_type) = 0; 147 const uint8_t payload_type) = 0;
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvideoengine2.cc | 76 int payload_type; member in struct:cricket::VideoCodecPref 181 VideoCodec default_codec(kDefaultVideoCodecPref.payload_type, 192 return VideoCodec(kRedPref.payload_type, kRedPref.name, 0, 0, 0, 0); 196 return VideoCodec(kUlpfecPref.payload_type, kUlpfecPref.name, 0, 0, 0, 0); 207 kDefaultVideoCodecPref.payload_type)); 984 config.encoder_settings.payload_type = codec_settings.codec.id; 988 config.rtp.rtx.payload_type = codec_settings.rtx_payload_type; 1690 int payload_type = in_codec.id; local [all...] |
webrtcvideoengine2_unittest.cc | 1102 int payload_type = kIncorrectPayloads[i]; local [all...] |
/external/chromium_org/chrome/common/ |
cast_messages.h | 51 IPC_STRUCT_TRAITS_MEMBER(payload_type)
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/external/chromium_org/media/cast/video_sender/ |
external_video_encoder_unittest.cc | 91 video_config_.rtp_config.payload_type = 127;
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/external/chromium_org/third_party/webrtc/video/ |
loopback.cc | 79 send_config.encoder_settings.payload_type = 124;
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call_tests.cc | 78 send_config_.encoder_settings.payload_type = kSendPayloadType; 87 decoder.payload_type = send_config_.encoder_settings.payload_type; 612 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; 613 int payload_type = send_config_.encoder_settings.payload_type; local 614 receive_config_.rtp.rtx[payload_type].ssrc = kSendRtxSsrc; 615 receive_config_.rtp.rtx[payload_type].payload_type = kSendRtxPayloadType; [all...] |
rampup_tests.cc | 455 send_config.encoder_settings.payload_type = 125; 468 send_config.rtp.rtx.payload_type = 96; 530 send_config.encoder_settings.payload_type = 125;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
neteq_impl.cc | 630 // We do not use |current_rtp_payload_type_| to |set payload_type|, but 638 int payload_type = rtp_header->payloadType; local 639 AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type); 642 decoder_database_->GetDecoderInfo(payload_type); 1128 int payload_type = packet->header.payloadType; local [all...] |