| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| mediasink.h | 33 // MediaSinkInterface is a sink to handle RTP and RTCP packets that are sent or
42 virtual void OnPacket(const void* data, size_t size, bool rtcp) = 0;
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| ssrcmuxfilter.h | 53 bool DemuxPacket(const char* data, size_t len, bool rtcp);
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| ssrcmuxfilter.cc | 49 bool SsrcMuxFilter::DemuxPacket(const char* data, size_t len, bool rtcp) {
51 if (!rtcp) {
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| channelmanager.h | 110 BaseSession* session, const std::string& content_name, bool rtcp);
116 BaseSession* session, const std::string& content_name, bool rtcp,
122 bool rtcp, DataChannelType data_channel_type);
250 BaseSession* session, const std::string& content_name, bool rtcp);
253 BaseSession* session, const std::string& content_name, bool rtcp,
258 bool rtcp, DataChannelType data_channel_type);
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| channel.cc | 353 static const char* PacketType(bool rtcp) {
354 return (!rtcp) ? "RTP" : "RTCP";
357 static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
360 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
382 const std::string& content_name, bool rtcp)
388 rtcp_(rtcp),
409 FlushRtcpMessages(); // Send any outstanding RTCP packets.
426 if (rtcp() && rtcp_transport_channel == NULL) {
585 // When using RTCP multiplexing we might get RTCP packets on the RT
587 bool rtcp = PacketIsRtcp(channel, data, len); local
[all...] |
| mediarecorder.h | 52 // RtpDumpSink implements MediaSinkInterface by dumping the RTP/RTCP packets to
63 virtual void OnPacket(const void* data, size_t size, bool rtcp);
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| channel.h | 75 const std::string& content_name, bool rtcp);
241 bool rtcp() const { return rtcp_; } function in class:cricket::BaseChannel
264 bool SendPacket(bool rtcp, talk_base::Buffer* packet);
265 virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
266 void HandlePacket(bool rtcp, talk_base::Buffer* packet);
282 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
285 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
363 const std::string& content_name, bool rtcp);
480 const std::string& content_name, bool rtcp,
589 bool rtcp);
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| channelmanager.cc | 316 BaseSession* session, const std::string& content_name, bool rtcp) {
319 session, content_name, rtcp));
323 BaseSession* session, const std::string& content_name, bool rtcp) {
332 session, content_name, rtcp);
362 BaseSession* session, const std::string& content_name, bool rtcp,
366 content_name, rtcp, voice_channel));
370 BaseSession* session, const std::string& content_name, bool rtcp,
383 session, content_name, rtcp, voice_channel);
414 bool rtcp, DataChannelType channel_type) {
417 rtcp, channel_type))
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| mediarecorder.cc | 77 void RtpDumpSink::OnPacket(const void* data, size_t size, bool rtcp) {
84 if (!rtcp) {
87 // TODO(whyuan): Enable recording RTCP.
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| srtpfilter.cc | 166 // differently in RTP/RTCP mux and non-mux modes.
168 // - In the non-muxed case, RTP and RTCP are keyed with different
578 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
581 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); // rtcp still 80
608 rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
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| /external/srtp/test/ |
| rtpw.c | 315 crypto_policy_set_rtcp_default(&policy.rtcp);
319 crypto_policy_set_rtcp_default(&policy.rtcp);
323 crypto_policy_set_rtcp_default(&policy.rtcp);
336 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
368 * specification, since RTCP authentication is required. However,
381 policy.rtcp.cipher_type = NULL_CIPHER;
382 policy.rtcp.cipher_key_len = 0;
383 policy.rtcp.auth_type = NULL_AUTH;
384 policy.rtcp.auth_key_len = 0
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| dtls_srtp_driver.c | 183 err = crypto_policy_set_from_profile_for_rtcp(&policy.rtcp, profile);
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| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| rtpdump.h | 49 // the actual RTP or RTCP packet.
67 RtpDumpPacket(const void* d, size_t s, uint32 elapsed, bool rtcp)
69 is_rtcp(rtcp) {
83 bool is_rtcp; // True if the data below is a RTCP packet.
84 std::vector<uint8> data; // The actual RTP or RTCP packet.
119 // handle both RTP dump and RTCP dump. We assume that the dump does not mix
120 // RTP packets and RTCP packets.
168 // Write a RTP or RTCP packet. The parameters data points to the packet and
193 uint32 elapsed, bool rtcp);
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| channelmanager.h | 106 BaseSession* session, const std::string& content_name, bool rtcp);
112 BaseSession* session, const std::string& content_name, bool rtcp,
167 BaseSession* session, const std::string& content_name, bool rtcp);
170 BaseSession* session, const std::string& content_name, bool rtcp,
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| channel.cc | 64 static const char* PacketType(bool rtcp) {
65 return (!rtcp) ? "RTP" : "RTCP";
68 static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
71 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
122 FlushRtcpMessages(); // Send any outstanding RTCP packets.
242 // When using RTCP multiplexing we might get RTCP packets on the RTP
243 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
244 bool rtcp = (channel == rtcp_transport_channel_ | local
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| channelmanager.cc | 66 bool rtcp, VoiceChannel* voice_channel)
69 rtcp(rtcp),
74 bool rtcp; member in struct:cricket::CreationParams
294 BaseSession* session, const std::string& content_name, bool rtcp) {
295 CreationParams params(session, content_name, rtcp, NULL);
300 BaseSession* session, const std::string& content_name, bool rtcp) {
311 session, content_name, rtcp);
338 BaseSession* session, const std::string& content_name, bool rtcp,
340 CreationParams params(session, content_name, rtcp, voice_channel)
[all...] |
| srtpfilter.cc | 362 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
365 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); // rtcp still 80
392 rtcp_auth_tag_len_ = policy.rtcp.auth_tag_len;
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| channel.h | 164 bool SendPacket(bool rtcp, talk_base::Buffer* packet);
165 void HandlePacket(bool rtcp, talk_base::Buffer* packet);
234 // Media sinks to handle the received or sent RTP/RTCP packets. These are
258 const std::string& content_name, bool rtcp);
369 const std::string& content_name, bool rtcp,
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| rtpdump.cc | 116 buf.ReadUInt16(&data_len); // data.size() for RTP, 0 for RTCP.
121 // Read the actual RTP or RTCP packet.
295 const void* data, size_t data_len, uint32 elapsed, bool rtcp) {
311 buf.WriteUInt16(static_cast<uint16>(rtcp ? 0 : data_len));
318 // Write the actual RTP or RTCP packet.
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| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump.h | 46 // the actual RTP or RTCP packet.
73 RtpDumpPacket(const void* d, size_t s, uint32 elapsed, bool rtcp)
75 original_data_len((rtcp) ? 0 : s) {
80 // In the rtpdump file format, RTCP packets have their data len set to zero,
81 // since RTCP has an internal length field.
92 // Get the type of the RTCP packet. Return true and set the output parameter
98 std::vector<uint8> data; // The actual RTP or RTCP packet.
141 // handle both RTP dump and RTCP dump. We assume that the dump does not mix
142 // RTP packets and RTCP packets.
192 // Write a RTP or RTCP packet. The parameters data points to the packet an
[all...] |
| rtpdump.cc | 141 // the header) was recorded. Note that this field is set to zero for RTCP
148 // Read the actual RTP or RTCP packet.
353 const void* data, size_t data_len, uint32 elapsed, bool rtcp) {
367 size_t write_len = FilterPacket(data, data_len, rtcp);
376 buf.WriteUInt16(static_cast<uint16>(rtcp ? 0 : data_len));
388 bool rtcp) {
390 if (!rtcp) {
403 // RTCP header + payload
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| testutils.cc | 136 size_t count, bool rtcp, uint32 rtp_ssrc, RtpDumpWriter* writer) {
143 if (rtcp) {
149 RtpDumpPacket dump_packet(buf.Data(), buf.Length(), elapsed_time_ms, rtcp);
175 // Check the RTP or RTCP packet.
179 // RTCP packet.
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| testutils.h | 98 // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return
101 size_t count, bool rtcp, uint32 rtp_ssrc, RtpDumpWriter* writer);
106 // payload. If the stream is a RTCP stream, verify the RTCP header and
|
| /external/srtp/include/ |
| srtp.h | 58 * @brief libSRTP provides functions for protecting RTP and RTCP. See
217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member in struct:srtp_policy_t
244 * RTCP destination transport addresses, using the RTP/SAVP (Secure
457 * structure to the SRTP default policy for RTCP protection.
462 * crypto_policy_t at location p to the SRTP default policy for RTCP
658 * structure to the appropriate value for RTCP based on an srtp_profile_t
663 * sets the crypto_policy_t at location policy to the policy for RTCP
719 * @defgroup SRTCP Secure RTCP
722 * @brief Secure RTCP functions are used to protect RTCP traffic
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| /external/srtp/srtp/ |
| srtp.c | 84 * This function allocates the stream context, rtp and rtcp ciphers
123 * ...and now the RTCP-specific initialization - first, allocate
126 stat = crypto_kernel_alloc_cipher(p->rtcp.cipher_type,
128 p->rtcp.cipher_key_len);
134 stat = crypto_kernel_alloc_auth(p->rtcp.auth_type,
136 p->rtcp.auth_key_len,
137 p->rtcp.auth_tag_len);
196 * deallocate rtcp cipher, if it is not the same as that in
206 * deallocate rtcp auth function, if it is not the same as that in
430 debug_print(mod_srtp, "found aes_icm, generating rtcp salt", NULL)
[all...] |
