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  /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
rtcp.h 23 class Rtcp {
25 Rtcp() {
29 ~Rtcp() {}
31 // Resets the RTCP statistics, and sets the first received sequence number.
34 // Updates the RTCP statistics with a new received packet.
37 // Returns the current RTCP statistics. If |no_reset| is true, the statistics
54 DISALLOW_COPY_AND_ASSIGN(Rtcp);
rtcp.cc 11 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
22 void Rtcp::Init(uint16_t start_sequence_number) {
33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
57 void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
neteq_impl.h 23 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
155 // Writes the current RTCP statistics to |stats|. The statistics are reset
365 Rtcp rtcp_ GUARDED_BY(crit_sect_);
  /external/chromium_org/media/cast/net/rtcp/
rtcp.cc 5 #include "media/cast/net/rtcp/rtcp.h"
12 #include "media/cast/net/rtcp/rtcp_builder.h"
13 #include "media/cast/net/rtcp/rtcp_defines.h"
14 #include "media/cast/net/rtcp/rtcp_utility.h"
57 Rtcp::Rtcp(const RtcpCastMessageCallback& cast_callback,
78 Rtcp::~Rtcp() {}
80 bool Rtcp::IsRtcpPacket(const uint8* packet, size_t length)
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rtcp.h 5 // This class maintains a bi-directional RTCP connection with a remote
25 #include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
26 #include "media/cast/net/rtcp/rtcp_builder.h"
27 #include "media/cast/net/rtcp/rtcp_defines.h"
52 class Rtcp {
54 Rtcp(const RtcpCastMessageCallback& cast_callback,
62 virtual ~Rtcp();
64 // Send a RTCP sender report.
76 // provided the RTCP receiver report will append a Cast message containing
78 // If |rtcp_events| is provided the RTCP receiver report will append th
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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
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  /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/
after_initialization_fixture.h 44 StorePacket(Packet::Rtcp, channel, data, len);
50 enum Type { Rtp, Rtcp, } type;
101 case Packet::Rtcp:
  /external/chromium_org/media/cast/net/
cast_transport_sender_impl.h 7 // Audio, video frames and RTCP messages are submitted to this object
12 // CastTransportSender RTP RTCP
15 // RtpSender (A/V) Rtcp (A/V)
19 // There are objects of TransportEncryptionHandler, RtpSender and Rtcp
21 // PacedSender and UdpTransport are shared between all RTP and RTCP
41 #include "media/cast/net/rtcp/rtcp.h"
128 // Called when a RTCP Cast message is received.
149 // Maintains RTCP session for audio and video.
150 scoped_ptr<Rtcp> audio_rtcp_session_
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cast_transport_sender_impl.cc 151 new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage,
184 new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage,
241 NOTREACHED() << "Invalid request for sending RTCP packet.";
357 VLOG(2) << "Received log message via RTCP that we did not expect: "
  /external/chromium_org/media/cast/receiver/
frame_receiver.h 17 #include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
18 #include "media/cast/net/rtcp/rtcp.h"
107 // Schedule timing for the next RTCP report.
113 // Actually send the next RTCP report.
128 // Processes raw events to be sent over to the cast sender via RTCP.
157 // Manages sending/receiving of RTCP packets, including sender/receiver
159 Rtcp rtcp_;
172 // it allows the event to be transmitted via RTCP.
182 // Time interval for sending a RTCP report
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cast_receiver_impl.cc 56 if (Rtcp::IsRtcpPacket(data, length)) {
57 ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length);
frame_receiver.cc 76 if (Rtcp::IsRtcpPacket(&packet->front(), packet->size())) {
  /external/chromium_org/media/cast/sender/
audio_sender_unittest.cc 30 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
33 // Check that at least one RTCP packet was sent before the first RTP
135 // Make sure that we send at least one RTCP packet.
video_sender_unittest.cc 66 // A singular packet implies a RTCP packet.
74 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
77 // Check that at least one RTCP packet was sent before the first RTP
304 // Make sure that we send at least one RTCP packet.
311 // Build Cast msg and expect RTCP packet.
  /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/
remote_bitrate_estimator_unittest_helper.h 79 // Generates an RTCP packet.
80 RtcpPacket* Rtcp(int64_t time_now_us);
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
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