| /external/chromium_org/media/audio/mac/ |
| D | audio_input_mac.cc | 190 UInt32 num_packets, in HandleInputBufferStatic()
201 UInt32 num_packets, in HandleInputBuffer()
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| /external/chromium_org/net/quic/ |
| D | quic_unacked_packet_map_test.cc | 39 size_t num_packets) { in VerifyInFlightPackets()
67 size_t num_packets) { in VerifyUnackedPackets()
82 size_t num_packets) { in VerifyRetransmittablePackets()
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| D | quic_time_wait_list_manager.h | 137 int num_packets; member
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| D | quic_fec_group_test.cc | 44 void RunTest(size_t num_packets, size_t lost_packet, bool out_of_order) { in RunTest()
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| D | quic_time_wait_list_manager.cc | 112 int num_packets = 0; in AddConnectionIdToTimeWait() local
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| D | quic_sent_packet_manager_test.cc | 74 size_t num_packets) { in VerifyUnackedPackets()
90 size_t num_packets) { in VerifyRetransmittablePackets()
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| /external/chromium_org/content/browser/speech/ |
| D | speech_recognizer_impl_unittest.cc | 415 int num_packets = (SpeechRecognizerImpl::kNoSpeechTimeoutMs) / in TEST_F() local
442 int num_packets = (SpeechRecognizerImpl::kNoSpeechTimeoutMs) / in TEST_F() local
479 int num_packets = SpeechRecognizerImpl::kEndpointerEstimationTimeMs / in TEST_F() local
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | fec_test_helper.cc | 20 void FrameGenerator::NewFrame(int num_packets) { in NewFrame()
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| D | rtp_format_video_generic.cc | 39 uint32_t num_packets = in SetPayloadData() local
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| /external/chromium_org/net/tools/quic/ |
| D | quic_time_wait_list_manager.h | 141 int num_packets; member
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| D | quic_time_wait_list_manager.cc | 116 int num_packets = 0; in AddConnectionIdToTimeWait() local
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| /external/chromium_org/media/cast/net/rtp/ |
| D | rtp_packetizer.cc | 50 size_t num_packets = (frame.data.size() + max_length) / max_length; in SendFrameAsPackets() local
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| D | receiver_stats_unittest.cc | 35 uint32 ExpectedJitter(uint32 const_interval, int num_packets) { in ExpectedJitter()
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| D | statistics_calculator.cc | 75 void StatisticsCalculator::PacketsDiscarded(int num_packets) { in PacketsDiscarded()
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| D | packet_buffer.cc | 259 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const { in BufferStat()
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| D | neteq_unittest.cc | 528 int num_packets = (frame_index % 10 == 0 ? 2 : 1); in TEST_F() local
559 int num_packets = (i % 10 == 9 ? 0 : 1); in TEST_F() local
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| /external/chromium_org/third_party/webrtc/video/ |
| D | replay.cc | 236 int num_packets = 0; in RtpReplay() local
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| /external/chromium_org/media/cast/logging/ |
| D | logging_impl_unittest.cc | 182 int num_packets = 0; in TEST_F() local
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| D | stats_event_subscriber_unittest.cc | 324 int num_packets = 10; in TEST_F() local
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| /external/chromium_org/media/audio/win/ |
| D | audio_low_latency_output_win.cc | 460 size_t num_packets = (num_available_frames / packet_size_frames_); in RenderAudioFromSource() local
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| /external/linux-tools-perf/perf-3.12.0/tools/perf/bench/ |
| D | sched-messaging.c | 47 unsigned int num_packets; member
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| /external/chromium_org/third_party/webrtc/modules/pacing/ |
| D | paced_sender_unittest.cc | 46 int num_packets = (bytes + kPaddingPacketSize - 1) / kPaddingPacketSize; in TimeToSendPadding() local
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| D | acm_receiver.cc | 764 int num_packets; in GetSilence() local
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| /external/qemu/android/ |
| D | shaper.c | 100 int num_packets; member
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| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| D | turnport_unittest.cc | 316 size_t num_packets = 256; in TestTurnSendData() local
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