| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_format_h264.h | 40 virtual bool NextPacket(uint8_t* buffer,
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| rtp_format_video_generic.cc | 47 bool RtpPacketizerGeneric::NextPacket(uint8_t* buffer,
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| rtp_format_vp8.h | 20 * After creating the packetizer, the method NextPacket is called
78 virtual bool NextPacket(uint8_t* buffer,
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| rtp_format_vp8_test_helper.cc | 79 EXPECT_TRUE(packetizer->NextPacket(buffer_, &send_bytes, &last));
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| rtp_format_h264.cc | 198 bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
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| rtp_sender_video.cc | 346 if (!packetizer->NextPacket(
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| fec_receiver_unittest.cc | 58 generator_->NextPacket(frame_offset + i, kRtpHeaderSize + 10));
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| video_rtp_play.cc | 61 while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
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| rtp_player.cc | 349 virtual int NextPacket(int64_t time_now) {
369 if (!packet_source_->NextPacket(&next_packet_))
394 if (!packet_source_->NextPacket(&next_packet_)) {
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| video_rtp_play_mt.cc | 48 if (rtp_player->NextPacket(clock->TimeInMilliseconds()) < 0) {
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| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| stunserver_unittest.cc | 68 rtc::TestClient::Packet* packet = client_->NextPacket();
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| relayserver_unittest.cc | 135 rtc::TestClient::Packet* packet = client->NextPacket();
146 rtc::TestClient::Packet* packet = client->NextPacket();
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_send_test.cc | 62 Packet* AcmSendTest::NextPacket() {
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| acm_send_test_oldapi.cc | 65 Packet* AcmSendTestOldApi::NextPacket() {
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| acm_receive_test.cc | 89 for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
90 packet.reset(packet_source_->NextPacket())) {
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| audio_coding_module_unittest.cc | 598 // AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
599 // packet from test::AcmSendTest::NextPacket, which inserts audio from the
688 virtual test::Packet* NextPacket() OVERRIDE {
691 test::Packet* packet = send_test_->NextPacket();
[all...] |
| audio_coding_module_unittest_oldapi.cc | 608 // AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
609 // packet from test::AcmSendTest::NextPacket, which inserts audio from the
701 test::Packet* NextPacket() OVERRIDE {
704 test::Packet* packet = send_test_->NextPacket();
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_analyze.cc | 95 packet.reset(file_source->NextPacket());
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| rtp_file_source.cc | 49 Packet* RtpFileSource::NextPacket() {
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| neteq_rtpplay.cc | 180 webrtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
199 next_packet.reset(file_source->NextPacket());
245 packet.reset(file_source->NextPacket());
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/tools/ |
| bwe_rtp_play.cc | 73 if (!rtp_reader->NextPacket(&packet)) {
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
| stream_generator.cc | 105 bool StreamGenerator::NextPacket(VCMPacket* packet) {
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| /external/chromium_org/third_party/webrtc/video/ |
| replay.cc | 240 if (!rtp_reader->NextPacket(&packet))
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| /external/chromium_org/third_party/webrtc/test/ |
| rtp_file_reader.cc | 102 virtual bool NextPacket(Packet* packet) OVERRIDE {
288 virtual bool NextPacket(Packet* packet) OVERRIDE {
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| jitter_buffer_unittest.cc | [all...] |
