| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| ssrcmuxfilter.cc | 50 uint32 ssrc = 0; local
52 GetRtpSsrc(data, len, &ssrc);
61 if (!GetRtcpSsrc(data, len, &ssrc)) return false;
62 if (ssrc == kSsrc01) {
63 // SSRC 1 has a special meaning and indicates generic feedback on
70 return FindStream(ssrc);
82 bool SsrcMuxFilter::RemoveStream(uint32 ssrc) {
83 return RemoveStreamBySsrc(&streams_, ssrc);
86 bool SsrcMuxFilter::FindStream(uint32 ssrc) const {
87 if (ssrc == 0)
[all...] |
| ssrcmuxfilter.h | 38 // This class maintains list of recv SSRC's destined for cricket::BaseChannel.
57 bool RemoveStream(uint32 ssrc);
59 bool FindStream(uint32 ssrc) const;
|
| channel.cc | 136 : ssrc(s),
141 uint32 ssrc; member in struct:cricket::PlayRingbackToneMessageData
148 DtmfMessageData(uint32 ssrc, int event, int duration, int flags)
149 : ssrc(ssrc),
155 uint32 ssrc; member in struct:cricket::DtmfMessageData
163 : ssrc(s),
168 uint32 ssrc; member in struct:cricket::ScaleVolumeMessageData
199 : ssrc(s), renderer(r), is_local(l), result(false) {}
200 uint32 ssrc; member in struct:cricket::AudioRenderMessageData
208 uint32 ssrc; member in struct:cricket::VideoRenderMessageData
218 uint32 ssrc; member in struct:cricket::AddScreencastMessageData
225 uint32 ssrc; member in struct:cricket::RemoveScreencastMessageData
234 uint32 ssrc; member in struct:cricket::ScreencastEventMessageData
253 uint32 ssrc; member in struct:cricket::VoiceChannelErrorMessageData
263 uint32 ssrc; member in struct:cricket::VideoChannelErrorMessageData
272 uint32 ssrc; member in struct:cricket::DataChannelErrorMessageData
285 uint32 ssrc; member in struct:cricket::SsrcMessageData
301 uint32 ssrc; member in struct:cricket::MuteStreamData
330 uint32 ssrc; member in struct:cricket::SetCapturerMessageData
347 uint32 ssrc; member in struct:cricket::VideoChannel::ScreencastDetailsMessageData
709 uint32 ssrc = 0; local
714 << ", seqnum=" << seq_num << ", SSRC=" << ssrc; local
805 uint32 ssrc = 0; local
810 << ", seqnum=" << seq_num << ", SSRC=" << ssrc; local
1971 uint32 ssrc; local
2427 uint32 ssrc = 0; local
[all...] |
| channel.h | 111 bool IsStreamMuted(uint32 ssrc);
121 // Mute sending media on the stream with SSRC |ssrc|
122 // If there is only one sending stream SSRC 0 can be used.
123 bool MuteStream(uint32 ssrc, bool mute);
127 bool RemoveRecvStream(uint32 ssrc);
129 bool RemoveSendStream(uint32 ssrc);
289 virtual bool MuteStream_w(uint32 ssrc, bool mute);
290 bool IsStreamMuted_w(uint32 ssrc);
294 bool RemoveRecvStream_w(uint32 ssrc);
[all...] |
| /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,
65 virtual bool SetCaptureDevice(uint32 ssrc,
67 // Enable/disable the video playout of a remote video track with |ssrc|.
68 virtual void SetVideoPlayout(uint32 ssrc, bool enable,
70 // Enable sending video on the local video track with |ssrc|.
71 virtual void SetVideoSend(uint32 ssrc, bool enable,
|
| mediastreamhandler.h | 50 TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc);
57 uint32 ssrc() const { return ssrc_; } function in class:webrtc::TrackHandler
76 uint32 ssrc,
97 uint32 ssrc,
117 uint32 ssrc,
137 uint32 ssrc,
160 virtual void AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) = 0;
161 virtual void AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) = 0;
183 uint32 ssrc) OVERRIDE;
185 uint32 ssrc) OVERRIDE
[all...] |
| mediastreamhandler.cc | 36 TrackHandler::TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc)
38 ssrc_(ssrc),
61 uint32 ssrc,
63 : TrackHandler(track, ssrc),
78 provider_->SetAudioSend(ssrc(), false, options, NULL);
87 provider_->SetAudioSend(ssrc(), audio_track_->enabled(), options,
93 uint32 ssrc,
95 : TrackHandler(track, ssrc),
105 provider_->SetAudioPlayout(ssrc(), false, NULL);
112 provider_->SetAudioPlayout(ssrc(), audio_track_->enabled()
[all...] |
| mediastreamsignaling.h | 68 uint32 ssrc) = 0;
73 uint32 ssrc) = 0;
86 uint32 ssrc) = 0;
91 uint32 ssrc) = 0;
125 // session description. This will set the SSRC used for sending data on
128 // session description. If the DataChannel label and a SSRC is included in
129 // the description, the DataChannel is updated with SSRC that will be used
131 // 4. When both the local and remote SSRC of a DataChannel is set the state of
136 // session description. If a label and a SSRC of a new DataChannel is found
137 // MediaStreamSignalingObserver::OnAddDataChannel with the label and SSRC i
288 uint32 ssrc; member in struct:webrtc::MediaStreamSignaling::TrackInfo
[all...] |
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
| rtp_packetizer_config.cc | 11 : ssrc(0),
|
| rtp_packetizer_config.h | 25 // SSRC.
26 unsigned int ssrc; member in struct:media::cast::RtpPacketizerConfig
|
| /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...] |
| hybridvideoengine.cc | 108 bool HybridVideoMediaChannel::SetRenderer(uint32 ssrc,
112 ret = channel1_->SetRenderer(ssrc, renderer);
115 ret = channel2_->SetRenderer(ssrc, renderer);
131 bool HybridVideoMediaChannel::MuteStream(uint32 ssrc, bool muted) {
134 ret = channel1_->MuteStream(ssrc, muted);
137 ret = channel2_->MuteStream(ssrc, muted);
175 bool HybridVideoMediaChannel::SetSendStreamFormat(uint32 ssrc,
177 return active_channel_ && active_channel_->SetSendStreamFormat(ssrc, format);
219 bool HybridVideoMediaChannel::SetCapturer(uint32 ssrc,
223 ret = channel1_->SetCapturer(ssrc, capturer)
[all...] |
| filemediaengine.h | 143 virtual bool RegisterVoiceProcessor(uint32 ssrc,
148 virtual bool UnregisterVoiceProcessor(uint32 ssrc,
206 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
209 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
219 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
222 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
226 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
229 virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) {
241 virtual bool RemoveSendStream(uint32 ssrc);
243 virtual bool RemoveRecvStream(uint32 ssrc) { return true;
[all...] |
| voiceprocessor.h | 50 virtual void OnFrame(uint32 ssrc,
|
| rtpdataengine.cc | 180 << "' with ssrc=" << stream.first_ssrc()
186 // TODO(pthatcher): This should be per-stream, not per-ssrc.
193 << "' with ssrc=" << stream.first_ssrc();
197 bool RtpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
199 if (!GetStreamBySsrc(send_streams_, ssrc, &found_stream)) {
203 RemoveStreamBySsrc(&send_streams_, ssrc);
204 delete rtp_clock_by_send_ssrc_[ssrc];
205 rtp_clock_by_send_ssrc_.erase(ssrc);
217 << "' with ssrc=" << stream.first_ssrc()
224 << "' with ssrc=" << stream.first_ssrc()
[all...] |
| /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));
33 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value));
40 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
42 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) {
47 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value));
49 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
52 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), &value))
193 const std::string& ssrc = *ssrc_iterator; local
[all...] |
| /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/srtp/include/ |
| srtp_priv.h | 80 uint32_t ssrc; /* synchronization source */ member in struct:__anon26591
94 uint32_t ssrc; /* synchronization source */ member in struct:__anon26592
120 uint32_t ssrc; /* synchronization source */ member in struct:__anon26594
139 uint32_t ssrc; /* synchronization source */ member in struct:__anon26596
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/media/cast/rtp_receiver/rtp_parser/ |
| rtp_parser.cc | 33 rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
37 // Not a valid payload type / ssrc combination.
52 uint32 rtp_timestamp, ssrc; local
56 big_endian_reader.ReadU32(&ssrc);
58 if (ssrc != parser_config_.ssrc) return false;
64 rtp_header->webrtc.header.ssrc = ssrc;
|
| rtp_parser.h | 18 ssrc = 0;
23 uint32 ssrc; member in struct:media::cast::RtpParserConfig
|
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.cc | 29 ssrc(0),
66 uint32 rtp_timestamp, ssrc; local
68 big_endian_reader.ReadU32(&ssrc);
76 parsed_packet->ssrc = ssrc;
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvoiceengine.h | 140 bool RegisterProcessor(uint32 ssrc,
143 bool UnregisterProcessor(uint32 ssrc,
212 uint32* ssrc) const;
213 bool FindChannelNumFromSsrc(uint32 ssrc,
221 uint32 ssrc,
273 // When the media processor registers with the engine, the ssrc is cached
340 virtual bool RemoveSendStream(uint32 ssrc);
342 virtual bool RemoveRecvStream(uint32 ssrc);
343 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
344 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer)
[all...] |
