/external/chromium_org/media/cast/net/rtp/ |
rtp_header_parser.cc | 31 num_csrcs(0), 56 const uint8 num_csrcs = rtp_data_begin_[0] & 0x0f; local 69 const uint8 csrc_octs = num_csrcs * 4; 76 parsed_packet->num_csrcs = num_csrcs; 78 parsed_packet->audio_num_energy = parsed_packet->num_csrcs;
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rtp_header_parser.h | 34 uint8 num_csrcs; member in struct:media::cast::RtpCastTestHeader
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rtp_packetizer_unittest.cc | 52 EXPECT_EQ(0, rtp_header.num_csrcs);
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/external/chromium_org/third_party/webrtc/video_engine/mock/ |
mock_vie_frame_provider_base.h | 23 int num_csrcs,
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/external/chromium_org/third_party/webrtc/video_engine/ |
vie_frame_provider_base.h | 32 int num_csrcs = 0, 79 int num_csrcs = 0,
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vie_frame_provider_base.cc | 49 int num_csrcs, 60 frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC); 65 (*it)->DeliverFrame(id_, video_frame, num_csrcs, CSRC); 72 (*it)->DeliverFrame(id_, extra_frame_.get(), num_csrcs, CSRC);
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vie_renderer.h | 106 int num_csrcs = 0,
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vie_encoder.cc | 454 int num_csrcs, 487 if (num_csrcs > 0) { 489 for (int i = 0; i < num_csrcs; i++) { 496 default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
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vie_encoder.h | 98 int num_csrcs = 0,
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vie_renderer.cc | 140 int num_csrcs,
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vie_channel.cc | 863 int num_csrcs = vie_receiver_.GetCsrcs(arrayCSRC); local 864 if (num_csrcs > 0) { 865 memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(uint32_t)); [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_receiver_impl.cc | 447 const uint8_t num_csrcs = rtp_header.header.numCSRCs; local 448 if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) { 452 num_csrcs * sizeof(uint32_t)); 454 if (num_csrcs > 0 || old_num_csrcs > 0) { 455 num_csrcs_diff = num_csrcs - old_num_csrcs; 456 num_csrcs_ = num_csrcs; // Update stored CSRCs.
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rtp_sender.cc | [all...] |