/frameworks/opt/net/voip/src/java/android/net/rtp/ |
AudioCodec.java | 51 public final String fmtp; field in class:AudioCodec 84 private AudioCodec(int type, String rtpmap, String fmtp) { 87 this.fmtp = fmtp; 103 * @param fmtp The format parameters specified in the corresponding SDP 107 public static AudioCodec getCodec(int type, String rtpmap, String fmtp) { 137 if (hint == AMR && fmtp != null) { 138 String clue = fmtp.toLowerCase(); 144 return new AudioCodec(type, rtpmap, fmtp);
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AudioGroup.java | 151 codec.rtpmap, codec.fmtp);
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/cts/tests/tests/net/src/android/net/rtp/cts/ |
AudioCodecTest.java | 23 private void assertEquals(AudioCodec codec, int type, String rtpmap, String fmtp) { 30 assertEquals(codec.fmtp, fmtp);
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/frameworks/opt/net/voip/src/jni/rtp/ |
AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp); 67 int AmrCodec::set(int sampleRate, const char *fmtp) 70 if (strcasestr(fmtp, "crc=1") || strcasestr(fmtp, "robust-sorting=1") || 71 strcasestr(fmtp, "interleaving=")) { 76 const char *modes = strcasestr(fmtp, "mode-set="); 93 mOctetAligned = (strcasestr(fmtp, "octet-align=1") != NULL); 211 int set(int sampleRate, const char *fmtp) {
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AudioCodec.h | 29 virtual int set(int sampleRate, const char *fmtp) = 0;
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GsmCodec.cpp | 42 int set(int sampleRate, const char *fmtp) {
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G711Codec.cpp | 37 int set(int sampleRate, const char *fmtp) { 88 int set(int sampleRate, const char *fmtp) {
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/frameworks/opt/net/voip/src/java/android/net/sip/ |
SimpleSessionDescription.java | 52 * a=fmtp:127 0-15 288 * Returns the {@code fmtp} attribute of the given format or 292 return super.get("a=fmtp:" + format, ' '); 296 * Sets a format and its {@code fmtp} attribute. If the attribute is 299 public void setFormat(String format, String fmtp) { 303 super.set("a=fmtp:" + format, ' ', fmtp); 307 * Removes a format and its {@code fmtp} attribute. 312 super.set("a=fmtp:" + format, ' ', null); 339 * Returns the {@code fmtp} attribute of the given RTP payload type o [all...] |
SipAudioCall.java | 742 media.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp); 771 reply.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp); 825 media.setRtpPayload(codec.type, codec.rtpmap, codec.fmtp); [all...] |
/frameworks/av/media/libstagefright/rtsp/ |
rtp_test.cpp | 95 "a=fmtp:97 packetization-mode=1;profile-level-id=42000C;" 128 "a=fmtp:97 octet-align\r\n"; 141 "a=fmtp:96 packetization-mode=1;profile-level-id=42001E;" 157 "a=fmtp:96 packetization-mode=1;profile-level-id=42001E;"
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ASessionDescription.cpp | 105 if (key == "a=fmtp" || key == "a=rtpmap" 219 sprintf(key, "a=fmtp:%lu", x);
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ARTPWriter.cpp | 511 "a=fmtp:" PT_STR " profile-level-id="); 520 sdp.append("a=fmtp:" PT_STR " octed-align\r\n");
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MyTransmitter.h | 203 "a=fmtp:" PT_STR " profile-level-id=42C015;sprop-parameter-sets=");
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
webrtcsdp.cc | 145 static const char kAttributeFmtp[] = "fmtp"; [all...] |
webrtcsdp_unittest.cc | [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
testsdpstrings.h | 52 "a=fmtp:101 0-15\r\n" 81 "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"
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/external/chromium_org/third_party/opus/src/doc/ |
draft-ietf-payload-rtp-opus.xml | 700 "usedtx", when present, MUST be included in the "a=fmtp" attribute 704 SSRC-specific "fmtp" source-level attribute (as defined in 708 and "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
constants.h | 60 // fmtp parameters
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/external/qemu/audio/ |
audio.c | 498 audfmt_e *fmtp = opt->valp; local 502 audio_audfmt_to_string (*fmtp) 585 audfmt_e *fmtp = opt->valp; local 586 *fmtp = audio_get_conf_fmt (optname, *fmtp, &def); [all...] |
/external/chromium_org/content/test/data/media/ |
peerconnection-call.html | 33 sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10'); [all...] |
/external/linux-tools-perf/perf-3.12.0/tools/perf/ |
builtin-trace.c | 350 static int syscall_fmt__cmp(const void *name, const void *fmtp) 352 const struct syscall_fmt *fmt = fmtp; [all...] |
/external/libvorbis/doc/ |
rfc5215.txt | [all...] |
rfc5215.xml | [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvoiceengine.cc | 475 // Only add fmtp parameters that differ from the spec. [all...] |
/external/robolectric/lib/main/ |
android.jar | |