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  /external/chromium_org/base/containers/
mru_cache.h 42 template <class KeyType, class PayloadType, class DeletorType,
48 typedef std::pair<KeyType, PayloadType> value_type;
90 iterator Put(const KeyType& key, const PayloadType& payload) {
213 template<class PayloadType>
216 void operator()(PayloadType& payload) {
222 template <class KeyType, class PayloadType>
224 PayloadType,
225 MRUCacheNullDeletor<PayloadType> > {
227 typedef MRUCacheBase<KeyType, PayloadType,
228 MRUCacheNullDeletor<PayloadType> > ParentType
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  /external/chromium_org/media/cast/rtp_receiver/rtp_parser/include/mock/
mock_rtp_feedback.h 17 int32(const int8 payloadType,
  /external/chromium_org/chrome/browser/extensions/api/cast_channel/
cast_channel.proto 44 enum PayloadType {
48 required PayloadType payload_type = 5;
  /external/smack/asmack-master/jingle/
60-remove-jingle_mediaimpl.patch 29 -import org.jivesoftware.smackx.jingle.media.PayloadType;
42 - * @param payloadType Payload of the jmf
47 - public TestMediaSession(final PayloadType payloadType, final TransportCandidate remote, final TransportCandidate local,
49 - super(payloadType, remote, local, "Test", jingleSession);
127 -import org.jivesoftware.smackx.jingle.media.PayloadType;
144 - private List<PayloadType> payloads = new ArrayList<PayloadType>();
146 - private PayloadType preferredPayloadType = null;
157 - public List<PayloadType> getPayloads()
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60-jingle-ext.patch 137 for (PayloadType payloadType : payloads) {
138 if (payloadType != null) {
  /external/chromium_org/media/cast/rtp_receiver/rtp_parser/
rtp_parser.cc 32 if (rtp_header->webrtc.header.payloadType == parser_config_.payload_type &&
61 rtp_header->webrtc.header.payloadType = payload_type;
rtp_parser_unittest.cc 45 EXPECT_EQ(kTestPayloadType, parsed_header.webrtc.header.payloadType);
  /external/chromium_org/media/cast/audio_receiver/
audio_decoder_unittest.cc 60 rtp_header.webrtc.header.payloadType = 127;
110 rtp_header.webrtc.header.payloadType = 127;
179 rtp_header.webrtc.header.payloadType = 127;
  /external/chromium_org/chrome/common/extensions/api/
cast_streaming_rtp_stream.idl 17 long payloadType;
  /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/
H264TrackImpl.java 563 int payloadType = 0;
593 payloadType = 0;
598 payloadType += last_payload_type_bytes;
602 payloadType += last_payload_type_bytes;
613 if (payloadType == 1) { // pic_timing is what we are interested in!
710 "payloadType=" + payloadType +
712 if (payloadType == 1) {
  /external/chromium_org/third_party/libjingle/source/talk/media/other/
linphonemediaengine.cc 228 int payloadtype = buf[1] & 0x7f; local
229 if (play_ && payloadtype != 13)
  /external/webrtc/src/modules/interface/
module_common_types.h 20 WebRtc_UWord8 payloadType;
315 payloadType = data.payloadType;
349 payloadType = data.payloadType;
382 WebRtc_UWord8 payloadType;
  /external/tcpdump/
sctpHeader.h 300 u_int32_t payloadtype; member in struct:sctpDataPart
print-sctp.c 194 printf("[PPID 0x%x] ", EXTRACT_32BITS(&dataHdrPtr->payloadtype));
  /frameworks/av/media/libstagefright/codecs/on2/h264dec/source/
h264bsd_sei.h 220 u32 payloadType;
h264bsd_sei.c 187 u32 tmp, payloadType, payloadSize, status;
199 payloadType = 0;
202 payloadType += 255;
206 payloadType += tmp;
217 pSeiMessage->payloadType = payloadType;
219 switch (payloadType)
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  /external/wpa_supplicant_8/src/rsn_supp/
tdls.c 76 u8 payloadtype; /* IEEE80211_TDLS_RFTYPE */ member in struct:wpa_tdls_frame
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  /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
fakewebrtcvoiceengine.h 775 WEBRTC_STUB(InsertExtraRTPPacket, (int channel, unsigned char payloadType,
    [all...]
webrtcvoiceengine.cc     [all...]
  /external/chromium_org/third_party/libjingle/source/talk/session/media/
mediasession.cc     [all...]

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