| /external/chromium_org/third_party/webrtc/system_wrappers/source/ |
| rtp_to_ntp_unittest.cc | 37 RtcpList rtcp; local
43 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
46 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
50 // This expected to fail since it's highly unlikely that the older RTCP
52 EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms));
56 RtcpList rtcp; local
62 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
65 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
67 EXPECT_TRUE(RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, ×tamp_in_ms))
76 RtcpList rtcp; local
96 RtcpList rtcp; local
113 RtcpList rtcp; local
133 RtcpList rtcp; local
[all...] |
| rtp_to_ntp.cc | 76 // This RTCP has already been added to the list.
81 // We need two RTCP SR reports to map between RTP and NTP. More than two will
92 // pairs in |rtcp|. The converted timestamp is returned in
96 const RtcpList& rtcp,
98 assert(rtcp.size() == 2);
99 int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
100 rtcp.front().ntp_frac);
101 int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs,
102 rtcp.back().ntp_frac);
103 int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp
[all...] |
| /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;
|
| bundlefilter.h | 46 // For rtp packets, this is decided based on the payload type. For rtcp packets,
54 bool DemuxPacket(const char* data, size_t len, bool rtcp);
|
| bundlefilter.cc | 43 bool BundleFilter::DemuxPacket(const char* data, size_t len, bool rtcp) {
45 // For rtcp packets, we check whether the ssrc can be found or is the special
47 // |streams_| is empty, we will allow all rtcp packets pass through provided
48 // that they are valid rtcp packets in case that they are for early media.
49 if (!rtcp) {
61 // Rtcp packets using ssrc filter.
78 // Pass through if |streams_| is empty to allow early rtcp packets in.
|
| channelmanager.h | 113 BaseSession* session, const std::string& content_name, bool rtcp);
119 BaseSession* session, const std::string& content_name, bool rtcp,
125 bool rtcp, DataChannelType data_channel_type);
262 BaseSession* session, const std::string& content_name, bool rtcp);
265 BaseSession* session, const std::string& content_name, bool rtcp,
270 bool rtcp, DataChannelType data_channel_type);
|
| channel.cc | 128 static const char* PacketType(bool rtcp) {
129 return (!rtcp) ? "RTP" : "RTCP";
132 static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) {
135 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
157 const std::string& content_name, bool rtcp)
163 rtcp_(rtcp),
185 FlushRtcpMessages(); // Send any outstanding RTCP packets.
202 if (rtcp() && rtcp_transport_channel == NULL) {
224 // Both RTP and RTCP channels are set, we can call SetInterface o
374 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);
|
| channel.h | 80 const std::string& content_name, bool rtcp);
248 bool rtcp() const { return rtcp_; } function in class:cricket::BaseChannel
270 bool SendPacket(bool rtcp, rtc::Buffer* packet,
272 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
273 void HandlePacket(bool rtcp, rtc::Buffer* packet,
292 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
295 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
403 const std::string& content_name, bool rtcp);
512 const std::string& content_name, bool rtcp,
607 bool rtcp);
[all...] |
| channelmanager.cc | 319 BaseSession* session, const std::string& content_name, bool rtcp) {
322 session, content_name, rtcp));
326 BaseSession* session, const std::string& content_name, bool rtcp) {
335 session, content_name, rtcp);
365 BaseSession* session, const std::string& content_name, bool rtcp,
369 content_name, rtcp, voice_channel));
373 BaseSession* session, const std::string& content_name, bool rtcp,
386 session, content_name, rtcp, voice_channel);
417 bool rtcp, DataChannelType channel_type) {
420 rtcp, channel_type))
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| stream_synchronization.h | 26 Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
27 RtcpList rtcp; member in struct:webrtc::StreamSynchronization::Measurements
|
| stream_synchronization.cc | 62 if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
63 // We need two RTCP SR reports per stream to do synchronization.
68 audio_measurement.rtcp,
74 video_measurement.rtcp,
|
| stream_synchronization_unittest.cc | 37 RtcpMeasurement rtcp; local
38 NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
39 rtcp.rtp_timestamp = NowRtp(frequency, offset);
40 return rtcp;
84 // Generates the necessary RTCP measurements and RTP timestamps and computes
103 // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP.
104 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
108 video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
112 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency
[all...] |
| vie_sync_module.cc | 47 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver_unittest.cc | 47 // Injects an RTCP packet into the receiver.
94 // Injects an RTCP packet into the receiver.
100 true); // Allow non-compound RTCP
153 rtcp::SenderReport sr;
155 rtcp::RawPacket p = sr.Build();
167 rtcp::SenderReport sr;
169 rtcp::RawPacket p = sr.Build();
177 rtcp::ReceiverReport rr;
179 rtcp::RawPacket p = rr.Build();
193 rtcp::ReportBlock rb
[all...] |
| rtcp_packet_unittest.cc | 18 using webrtc::rtcp::App;
19 using webrtc::rtcp::Bye;
20 using webrtc::rtcp::Dlrr;
21 using webrtc::rtcp::Empty;
22 using webrtc::rtcp::Fir;
23 using webrtc::rtcp::Ij;
24 using webrtc::rtcp::Nack;
25 using webrtc::rtcp::Pli;
26 using webrtc::rtcp::Sdes;
27 using webrtc::rtcp::SenderReport
[all...] |
| /external/chromium_org/third_party/webrtc/system_wrappers/interface/ |
| rtp_to_ntp.h | 30 // Updates |rtcp_list| with timestamps from the latest RTCP SR.
41 bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
|
| /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
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump.h | 47 // the actual RTP or RTCP packet.
74 RtpDumpPacket(const void* d, size_t s, uint32 elapsed, bool rtcp)
76 original_data_len((rtcp) ? 0 : s) {
81 // In the rtpdump file format, RTCP packets have their data len set to zero,
82 // since RTCP has an internal length field.
93 // Get the type of the RTCP packet. Return true and set the output parameter
99 std::vector<uint8> data; // The actual RTP or RTCP packet.
142 // handle both RTP dump and RTCP dump. We assume that the dump does not mix
143 // RTP packets and RTCP packets.
193 // 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
|
| /external/chromium_org/third_party/libsrtp/srtp/test/ |
| rtpw.c | 335 crypto_policy_set_rtcp_default(&policy.rtcp);
339 crypto_policy_set_rtcp_default(&policy.rtcp);
343 crypto_policy_set_rtcp_default(&policy.rtcp);
357 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
389 * specification, since RTCP authentication is required. However,
402 policy.rtcp.cipher_type = NULL_CIPHER;
403 policy.rtcp.cipher_key_len = 0;
404 policy.rtcp.auth_type = NULL_AUTH;
405 policy.rtcp.auth_key_len = 0
[all...] |
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.cc | 79 // Generates an RTCP packet.
80 RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
84 RtcpPacket* rtcp = new RtcpPacket; local
86 rtcp->timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
88 rtcp->ntp_secs = send_time_us / 1000000;
89 rtcp->ntp_frac = static_cast<int64_t>((send_time_us % 1000000) *
91 rtcp->ssrc = ssrc_;
93 return rtcp;
206 0)); // RTCP receive time.
456 0)); // RTCP receive time
[all...] |
| /external/chromium_org/third_party/libsrtp/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
466 * structure to the SRTP default policy for RTCP protection.
471 * crypto_policy_t at location p to the SRTP default policy for RTCP
731 * structure to the appropriate value for RTCP based on an srtp_profile_t
736 * sets the crypto_policy_t at location policy to the policy for RTCP
792 * @defgroup SRTCP Secure RTCP
795 * @brief Secure RTCP functions are used to protect RTCP traffic
[all...] |
| /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
[all...] |
| /external/chromium_org/media/cast/net/rtcp/ |
| rtcp_unittest.cc | 11 #include "media/cast/net/rtcp/rtcp.h"
30 void set_rtcp_destination(Rtcp* rtcp) { rtcp_ = rtcp; }
56 Rtcp* rtcp_;
132 Rtcp rtcp_for_sender_;
133 Rtcp rtcp_for_receiver_;
141 // received a RTCP packet.
169 // need to fill-in more testing of RTCP now that much of the refactoring wor
[all...] |
