| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| decode_bwe.c | 29 WebRtc_Word32 packet_size,
59 (WebRtc_Word16) packet_size, /* in bytes */
|
| isacfix.c | 555 * - packet_size : size of the packet.
566 WebRtc_Word32 packet_size,
585 if (packet_size <= 0) {
589 } else if (packet_size > (STREAM_MAXW16<<1)) {
614 if (packet_size == 0)
623 packet_size,
648 * - packet_size : size of the packet.
660 WebRtc_Word32 packet_size,
680 if (packet_size <= 0) {
684 } else if (packet_size > (STREAM_MAXW16<<1))
[all...] |
| /external/eigen/test/eigen2/ |
| eigen2_first_aligned.cpp | 15 const int packet_size = sizeof(Scalar) * ei_packet_traits<Scalar>::size; local
16 VERIFY(((std::size_t(array) + sizeof(Scalar) * ei_alignmentOffset(array, size)) % packet_size) == 0);
|
| /external/eigen/test/ |
| first_aligned.cpp | 15 const int packet_size = sizeof(Scalar) * internal::packet_traits<Scalar>::size; local
16 VERIFY(((size_t(array) + sizeof(Scalar) * internal::first_aligned(array, size)) % packet_size) == 0);
|
| /external/chromium_org/net/quic/congestion_control/ |
| fix_rate_test.cc | 89 const QuicByteCount packet_size = 1200; local
103 sender_->OnPacketSent(clock_.Now(), sequence_number++, packet_size,
108 sender_->OnPacketSent(clock_.Now(), sequence_number++, packet_size,
113 sender_->OnPacketAcked(sequence_number - 1, packet_size, rtt_);
114 sender_->OnPacketAcked(sequence_number - 2, packet_size, rtt_);
117 EXPECT_EQ(num_packets * packet_size * 1000000 / bitrate.ToBytesPerSecond(),
|
| available_channel_estimator.h | 42 QuicByteCount packet_size,
|
| channel_estimator.h | 39 QuicByteCount packet_size,
|
| available_channel_estimator.cc | 27 QuicByteCount packet_size,
47 received_bytes_ += packet_size;
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| channel_estimator.cc | 32 QuicByteCount packet_size,
48 UpdateFilter(received_delta, packet_size, sequence_number);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| decode_bwe.c | 21 WebRtc_Word32 packet_size,
80 arrivalTimestampIn16kHz, packet_size);
|
| codec.h | 28 WebRtc_Word32 packet_size,
|
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host_tcp.cc | 422 int packet_size = base::NetToHost16(*reinterpret_cast<uint16*>(input)); local
423 if (input_len < packet_size + kPacketHeaderSize)
428 std::vector<char> data(cur, cur + packet_size);
430 consumed += packet_size;
463 int packet_size = GetExpectedPacketSize( local
466 if (input_len < packet_size + pad_bytes)
472 std::vector<char> data(cur, cur + packet_size);
474 consumed += packet_size;
519 int packet_size = base::NetToHost16(*reinterpret_cast<const uint16*>( local
528 packet_size += kStunHeaderSize
[all...] |
| /external/chromium_org/remoting/codec/ |
| audio_encoder_opus_unittest.cc | 127 void TestEncodeDecode(int packet_size,
137 for (; pos < kTotalTestSamples; pos += packet_size) {
139 CreatePacket(packet_size, rate, frequency_hz, pos);
160 ValidateReceivedData(packet_size, kDefaultSamplingRate,
|
| /external/chromium_org/media/audio/pulse/ |
| pulse_input.cc | 274 int packet_size = params_.GetBytesPerBuffer(); local
275 while (buffer_->forward_bytes() >= packet_size) {
276 buffer_->Read(audio_data_buffer_.get(), packet_size);
277 callback_->OnData(this, audio_data_buffer_.get(), packet_size,
280 if (buffer_->forward_bytes() < packet_size)
|
| pulse_unified.cc | 201 int packet_size = params_.GetBytesPerBuffer(); local
202 while (fifo_->forward_bytes() >= packet_size) {
203 WriteData(packet_size);
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| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/ |
| isacfix.h | 195 * - packet_size : size of the packet.
206 WebRtc_Word32 packet_size,
218 * - packet_size : size of the packet.
231 WebRtc_Word32 packet_size,
|
| /external/chromium_org/third_party/mesa/src/src/mapi/glapi/gen/ |
| glX_doc.py | 63 def packet_size(self): member in class:glx_doc_parameter
107 [s, pad] = p.packet_size()
208 [s, pad] = output.packet_size()
223 [s, pad] = p.packet_size()
|
| /external/mesa3d/src/mapi/glapi/gen/ |
| glX_doc.py | 63 def packet_size(self): member in class:glx_doc_parameter
107 [s, pad] = p.packet_size()
208 [s, pad] = output.packet_size()
223 [s, pad] = p.packet_size()
|
| /external/bluetooth/bluedroid/stack/rfcomm/ |
| port_utils.c | 145 UINT16 packet_size; local
151 packet_size = btm_get_max_packet_size (p_port->bd_addr);
152 if (packet_size == 0)
168 if ((L2CAP_MTU_SIZE + L2CAP_PKT_OVERHEAD) >= packet_size)
170 p_port->mtu = ((L2CAP_MTU_SIZE + L2CAP_PKT_OVERHEAD) / packet_size * packet_size) - RFCOMM_DATA_OVERHEAD - L2CAP_PKT_OVERHEAD;
|
| /external/chromium_org/content/renderer/media/ |
| media_stream_audio_processor_unittest.cc | 101 const int packet_size = local
103 const size_t length = packet_size * kNumberOfPacketsForTest;
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
| Isac_test.cc | 34 int packet_size, /* bytes */
46 BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
|
| /external/chromium/net/spdy/ |
| spdy_test_util.cc | 772 int packet_size = 0; local
789 packet_size += AppendToBuffer(header_string,
793 packet_size += AppendToBuffer(": ",
806 packet_size += AppendToBuffer(value_string + offset,
810 packet_size += AppendToBuffer("\n",
818 packet_size += AppendToBuffer(header_string,
822 packet_size += AppendToBuffer(": ",
829 packet_size += AppendToBuffer(value_string + offset,
833 packet_size += AppendToBuffer("\n",
838 return packet_size;
[all...] |
| /external/chromium_org/media/audio/alsa/ |
| alsa_output.cc | 366 size_t packet_size = frames_filled * bytes_per_frame_; local
367 DCHECK_LE(packet_size, packet_size_);
376 packet_size = packet_size / bytes_per_frame_ * bytes_per_output_frame_;
385 if (packet_size > 0) {
386 packet->set_data_size(packet_size);
|
| /external/chromium_org/net/quic/ |
| quic_connection_logger.cc | 24 size_t packet_size,
29 dict->SetInteger("size", packet_size);
36 size_t packet_size,
43 dict->SetInteger("size", packet_size);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/interface/ |
| isac.h | 181 * - packet_size : size of the packet.
194 WebRtc_Word32 packet_size,
|
