| /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);
|
| AudioGroup.java | 151 codec.rtpmap, codec.fmtp);
|
| /frameworks/opt/net/voip/src/jni/rtp/ |
| AudioCodec.h | 29 virtual int set(int sampleRate, const char *fmtp) = 0;
|
| 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) {
|
| GsmCodec.cpp | 42 int set(int sampleRate, const char *fmtp) {
|
| G711Codec.cpp | 37 int set(int sampleRate, const char *fmtp) {
88 int set(int sampleRate, const char *fmtp) {
|
| /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);
|
| /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...] |
| /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/third_party/libjingle/source/talk/app/webrtc/ |
| webrtcsdp.cc | 146 static const char kAttributeFmtp[] = "fmtp";
[all...] |
| /prebuilts/sdk/12/ |
| android.jar | |
| /prebuilts/sdk/14/ |
| android.jar | |
| /prebuilts/sdk/18/ |
| android.jar | |
| /prebuilts/sdk/19/ |
| android.jar | |
| /prebuilts/sdk/current/ |
| android.jar | |
