| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| mediastreamprovider.h | 47 // Enable/disable the audio playout of a remote audio track with |ssrc|.
48 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
50 // Enable/disable sending audio on the local audio track with |ssrc|.
52 virtual void SetAudioSend(uint32 ssrc, bool enable,
56 // Sets the audio playout volume of a remote audio track with |ssrc|.
58 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) = 0;
69 virtual bool SetCaptureDevice(uint32 ssrc,
71 // Enable/disable the video playout of a remote video track with |ssrc|.
72 virtual void SetVideoPlayout(uint32 ssrc, bool enable,
74 // Enable sending video on the local video track with |ssrc|
[all...] |
| mediastreamhandler.h | 51 TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc);
58 uint32 ssrc() const { return ssrc_; } function in class:webrtc::TrackHandler
99 uint32 ssrc,
125 uint32 ssrc,
148 uint32 ssrc,
168 uint32 ssrc,
191 virtual void AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) = 0;
192 virtual void AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) = 0;
214 uint32 ssrc) OVERRIDE;
216 uint32 ssrc) OVERRIDE
[all...] |
| mediastreamhandler.cc | 36 TrackHandler::TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc)
38 ssrc_(ssrc),
87 uint32 ssrc,
89 : TrackHandler(track, ssrc),
107 provider_->SetAudioSend(ssrc(), false, options, NULL);
124 provider_->SetAudioSend(ssrc(), audio_track_->enabled(), options, renderer);
129 uint32 ssrc,
131 : TrackHandler(track, ssrc),
143 provider_->SetAudioPlayout(ssrc(), false, NULL);
150 provider_->SetAudioPlayout(ssrc(), audio_track_->enabled()
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| ssrc_database.cc | 58 uint32_t ssrc = GenerateRandom(); local
62 while(_ssrcMap.find(ssrc) != _ssrcMap.end())
64 ssrc = GenerateRandom();
66 _ssrcMap[ssrc] = 0;
86 if (_ssrcVector[i] == ssrc)
89 i = 0; // start over with a new ssrc
90 ssrc = GenerateRandom();
95 _ssrcVector[_numberOfSSRC] = ssrc;
99 return ssrc;
103 SSRCDatabase::RegisterSSRC(const uint32_t ssrc)
201 uint32_t ssrc = 0; local
[all...] |
| rtcp_packet.h | 100 // | SSRC_1 (SSRC of first source) |
122 void To(uint32_t ssrc) {
123 report_block_.SSRC = ssrc;
156 // | SSRC of sender |
179 void From(uint32_t ssrc) {
180 sr_.SenderSSRC = ssrc;
225 // | SSRC of packet sender |
238 void From(uint32_t ssrc) {
239 rr_.SenderSSRC = ssrc;
337 uint32_t ssrc; member in struct:webrtc::rtcp::Sdes::Chunk
[all...] |
| ssrc_database.h | 31 int32_t RegisterSSRC(const uint32_t ssrc);
32 int32_t ReturnSSRC(const uint32_t ssrc);
41 // Friend function to allow the SSRC destructor to be accessed from the
|
| /external/chromium_org/chrome/browser/media/ |
| webrtc_browsertest_perf.cc | 14 // A ssrc stats key will be on the form stats.<bucket>-<key>.values.
24 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
26 if (!pc_dict.GetString(Statistic("audioOutputLevel", ssrc), &value)) {
31 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesReceived", ssrc), &value));
34 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value));
42 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
44 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) {
49 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value));
52 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
55 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), &value))
231 const std::string& ssrc = *ssrc_iterator; local
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| encoder_state_feedback.cc | 30 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
31 owner_->OnReceivedIntraFrameRequest(ssrc);
33 virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) {
34 owner_->OnReceivedSLI(ssrc, picture_id);
36 virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) {
37 owner_->OnReceivedRPSI(ssrc, picture_id);
56 bool EncoderStateFeedback::AddEncoder(uint32_t ssrc, ViEEncoder* encoder) {
58 if (encoders_.find(ssrc) != encoders_.end()) {
59 // Two encoders must not have the same ssrc.
63 encoders_[ssrc] = encoder
[all...] |
| encoder_state_feedback.h | 37 // Adds an encoder to receive feedback for a unique ssrc.
38 bool AddEncoder(uint32_t ssrc, ViEEncoder* encoder);
49 void OnReceivedIntraFrameRequest(uint32_t ssrc);
50 void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id);
51 void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id);
62 // Maps a unique ssrc to the given encoder.
|
| vie_remb_unittest.cc | 50 unsigned int ssrc = 1234; local
51 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
75 unsigned int ssrc = 1234; local
76 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
101 unsigned int ssrc[] = { 1234, 5678 }; local
102 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]))
132 unsigned int ssrc[] = { 1234, 5678 }; local
166 unsigned int ssrc[] = { 1234, 5678 }; local
200 unsigned int ssrc = 1234; local
233 unsigned int ssrc = 1234; local
[all...] |
| encoder_state_feedback_unittest.cc | 37 void(uint32_t ssrc, uint8_t picture_id));
39 void(uint32_t ssrc, uint64_t picture_id));
57 const int ssrc = 1234; local
59 EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
61 EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc))
64 OnReceivedIntraFrameRequest(ssrc);
67 EXPECT_CALL(encoder, OnReceivedSLI(ssrc, sli_picture_id))
70 ssrc, sli_picture_id);
73 EXPECT_CALL(encoder, OnReceivedRPSI(ssrc, rpsi_picture_id))
76 ssrc, rpsi_picture_id)
131 const int ssrc = 1234; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| fakemediaprocessor.h | 48 virtual void OnFrame(uint32 ssrc,
53 virtual void OnFrame(uint32 ssrc, VideoFrame* frame_ptr, bool* drop_frame) {
60 virtual void OnVoiceMute(uint32 ssrc, bool muted) {}
61 virtual void OnVideoMute(uint32 ssrc, bool muted) {}
|
| videoprocessor.h | 46 virtual void OnFrame(uint32 ssrc, VideoFrame* frame, bool* drop_frame) = 0;
|
| streamparams.h | 35 // Let the simulcast elements have SSRC 10, 20, 30.
37 // SSRC 11,21,31.
39 // StreamParams would then contain ssrc = {10,11,20,21,30,31} and
80 static StreamParams CreateLegacy(uint32 ssrc) {
82 stream.ssrcs.push_back(ssrc);
110 bool has_ssrc(uint32 ssrc) const {
111 return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end();
113 void add_ssrc(uint32 ssrc) {
114 ssrcs.push_back(ssrc);
132 // Convenience function to add an FID ssrc for a primary_ssr
189 uint32 ssrc; member in struct:cricket::StreamSelector
[all...] |
| fakemediaengine.h | 124 virtual bool RemoveSendStream(uint32 ssrc) {
125 return RemoveStreamBySsrc(&send_streams_, ssrc);
135 virtual bool RemoveRecvStream(uint32 ssrc) {
136 return RemoveStreamBySsrc(&receive_streams_, ssrc);
138 virtual bool MuteStream(uint32 ssrc, bool on) {
139 if (!HasSendStream(ssrc) && ssrc != 0)
142 muted_streams_.insert(ssrc);
144 muted_streams_.erase(ssrc);
147 bool IsStreamMuted(uint32 ssrc) const
231 uint32 ssrc; member in struct:cricket::FakeVoiceMediaChannel::DtmfInfo
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| bundlefilter.cc | 45 // For rtcp packets, we check whether the ssrc can be found or is the special
57 // Rtcp packets using ssrc filter.
59 uint32 ssrc = 0; local
66 if (!GetRtcpSsrc(data, len, &ssrc)) return false;
67 if (ssrc == kSsrc01) {
68 // SSRC 1 has a special meaning and indicates generic feedback on
75 return !HasStreams() || FindStream(ssrc);
91 bool BundleFilter::RemoveStream(uint32 ssrc) {
92 return RemoveStreamBySsrc(&streams_, ssrc);
99 bool BundleFilter::FindStream(uint32 ssrc) const
[all...] |
| bundlefilter.h | 47 // this is decided based on the sender ssrc values.
63 bool RemoveStream(uint32 ssrc);
68 bool FindStream(uint32 ssrc) const;
|
| /external/chromium_org/content/browser/resources/media/ |
| ssrc_info_manager.js | 8 * Get the ssrc if |report| is an ssrc report.
14 * @return {?string} The ssrc.
17 if (report.type != 'ssrc') {
18 console.warn("Trying to get ssrc from non-ssrc report.");
22 // If the 'ssrc' name-value pair exists, return the value; otherwise, return
24 // The 'ssrc' name-value pair only exists in an upcoming Libjingle change. Old
25 // versions use id to refer to the ssrc.
31 if (report.stats.values[i] == 'ssrc') {
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| send_statistics_proxy.cc | 58 StreamStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
59 std::map<uint32_t, StreamStats>::iterator it = stats_.substreams.find(ssrc);
63 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) ==
67 return &stats_.substreams[ssrc]; // Insert new entry and return ptr.
71 uint32_t ssrc) {
73 StreamStats* stats = GetStatsEntry(ssrc);
82 uint32_t ssrc) {
84 StreamStats* stats = GetStatsEntry(ssrc);
92 uint32_t ssrc) {
94 StreamStats* stats = GetStatsEntry(ssrc);
[all...] |
| send_statistics_proxy.h | 53 uint32_t ssrc) OVERRIDE;
56 uint32_t ssrc) OVERRIDE;
59 virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) OVERRIDE;
64 const unsigned int ssrc) OVERRIDE;
84 StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| /external/chromium_org/third_party/webrtc/modules/pacing/ |
| paced_sender_unittest.cc | 28 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms,
38 bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
67 uint32_t ssrc, uint16_t sequence_number,
70 EXPECT_FALSE(send_bucket_->SendPacket(priority, ssrc,
73 ssrc, sequence_number, capture_time_ms, false))
83 uint32_t ssrc = 12345; local
86 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
88 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
90 SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
93 EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
118 uint32_t ssrc = 12345; local
158 uint32_t ssrc = 12345; local
208 uint32_t ssrc = 12345; local
252 uint32_t ssrc = 12345; local
270 uint32_t ssrc = 12345; local
296 uint32_t ssrc = 12346; local
345 uint32_t ssrc = 12346; local
413 uint32_t ssrc = 12346; local
473 uint32_t ssrc = 12346; local
502 uint32_t ssrc = 12346; local
[all...] |
| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon16481
94 uint32_t ssrc; /* synchronization source */ member in struct:__anon16482
120 uint32_t ssrc; /* synchronization source */ member in struct:__anon16484
139 uint32_t ssrc; /* synchronization source */ member in struct:__anon16486
165 * srtp_get_stream(ssrc) returns a pointer to the stream corresponding
166 * to ssrc, or NULL if no stream exists for that ssrc
170 srtp_get_stream(srtp_t srtp, uint32_t ssrc);
202 * an srtp_stream_t has its own SSRC, encryption key, authentication
210 uint32_t ssrc; member in struct:srtp_stream_ctx_t
[all...] |
| /external/srtp/include/ |
| srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon34934
94 uint32_t ssrc; /* synchronization source */ member in struct:__anon34935
120 uint32_t ssrc; /* synchronization source */ member in struct:__anon34937
139 uint32_t ssrc; /* synchronization source */ member in struct:__anon34939
165 * srtp_get_stream(ssrc) returns a pointer to the stream corresponding
166 * to ssrc, or NULL if no stream exists for that ssrc
170 srtp_get_stream(srtp_t srtp, uint32_t ssrc);
209 * an srtp_stream_t has its own SSRC, encryption key, authentication
217 uint32_t ssrc; member in struct:srtp_stream_ctx_t
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_generator.h | 28 uint32_t ssrc = 0x12345678)
32 ssrc_(ssrc),
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.h | 53 unsigned int ssrc; member in struct:webrtc::testing::RtpStream::RtpPacket
60 unsigned int ssrc; member in struct:webrtc::testing::RtpStream::RtcpPacket
67 RtpStream(int fps, int bitrate_bps, unsigned int ssrc, unsigned int frequency,
86 unsigned int ssrc() const;
124 // Set the RTP timestamp offset for the stream identified by |ssrc|.
125 void set_rtp_timestamp_offset(unsigned int ssrc, uint32_t offset);
167 void IncomingPacket(uint32_t ssrc,
174 // with a given ssrc. The stream is pushed through a very simple simulated
179 bool GenerateAndProcessFrame(unsigned int ssrc, unsigned int bitrate_bps);
185 unsigned int SteadyStateRun(unsigned int ssrc,
[all...] |
