| /external/chromium_org/third_party/webrtc/test/testsupport/ |
| packet_reader_unittest.cc | 44 EXPECT_EQ(-1, reader_->NextPacket(&data_pointer));
50 ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_));
61 int length_to_read = reader_->NextPacket(&data_pointer);
66 length_to_read = reader_->NextPacket(&data_pointer);
78 int length_to_read = reader_->NextPacket(&data_pointer);
83 length_to_read = reader_->NextPacket(&data_pointer);
93 int length_to_read = reader_->NextPacket(&packet_data_pointer_);
97 length_to_read = reader_->NextPacket(&packet_data_pointer_);
101 length_to_read = reader_->NextPacket(&packet_data_pointer_);
106 length_to_read = reader_->NextPacket(&packet_data_pointer_)
[all...] |
| packet_reader.h | 26 // NextPacket method.
29 // * packet_size_in_bytes is the number of bytes to read in each NextPacket
40 virtual int NextPacket(uint8_t** packet_pointer);
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| packet_reader.cc | 37 int PacketReader::NextPacket(uint8_t** packet_pointer) {
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| packet_source.h | 32 virtual Packet* NextPacket() = 0;
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| rtp_file_source.h | 42 virtual Packet* NextPacket();
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| /external/chromium_org/third_party/webrtc/test/ |
| rtp_file_reader.h | 38 virtual bool NextPacket(Packet* packet) = 0;
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| rtp_file_reader_unittest.cc | 34 while (rtp_packet_source_->NextPacket(&packet))
63 while (rtp_packet_source_->NextPacket(&packet))
71 while (rtp_packet_source_->NextPacket(&packet)) {
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| fec_test_helper.h | 43 RtpPacket* NextPacket(int offset, size_t length);
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| rtp_format.h | 42 virtual bool NextPacket(uint8_t* buffer,
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| rtp_format_video_generic.h | 43 virtual bool NextPacket(uint8_t* buffer,
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| rtp_format_h264_unittest.cc | 100 ASSERT_TRUE(packetizer->NextPacket(packet.get(), &length, &last));
106 EXPECT_FALSE(packetizer->NextPacket(packet.get(), &length, &last));
171 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
176 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
201 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
205 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
210 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
238 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last));
246 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last));
273 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last))
[all...] |
| producer_fec_unittest.cc | 71 RtpPacket* rtp_packet = generator_->NextPacket(i, 10);
113 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10);
140 RtpPacket* packet = generator_->NextPacket(0, 10);
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| fec_test_helper.cc | 27 RtpPacket* FrameGenerator::NextPacket(int offset, size_t length) {
66 RtpPacket* red_packet = NextPacket(0, packet->length + 1);
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| packet_buffer_unittest.cc | 30 Packet* NextPacket(int payload_size_bytes);
47 Packet* PacketGenerator::NextPacket(int payload_size_bytes) {
83 Packet* packet = gen.NextPacket(payload_len);
106 Packet* packet = gen.NextPacket(payload_len);
127 Packet* packet = gen.NextPacket(payload_len);
136 Packet* packet = gen.NextPacket(payload_len);
156 Packet* packet = gen.NextPacket(payload_len);
192 Packet* packet = gen.NextPacket(payload_len);
196 Packet* packet = gen.NextPacket(payload_len);
258 Packet* packet = gen.NextPacket(payload_len)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_send_test.h | 44 virtual Packet* NextPacket() OVERRIDE;
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| acm_send_test_oldapi.h | 46 Packet* NextPacket();
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| acm_receive_test_oldapi.cc | 147 for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
148 packet.reset(packet_source_->NextPacket())) {
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| /external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/ |
| packet_loss_test.h | 37 virtual int NextPacket(int size, unsigned char **pkg);
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| packet_loss_test.cc | 195 while ((size = NextPacket(1500, &packet)) > 0)
230 int PacketLossTest::NextPacket(int mtu, unsigned char **pkg)
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| /external/chromium_org/third_party/webrtc/base/ |
| testclient.cc | 19 // NextPacket.
54 TestClient::Packet* TestClient::NextPacket() {
91 Packet* packet = NextPacket();
103 Packet* packet = NextPacket();
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| testclient.h | 60 Packet* NextPacket();
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/tools/ |
| rtp_to_text.cc | 47 while (rtp_reader->NextPacket(&packet)) {
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| /external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/ |
| packet_manipulator.cc | 53 while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
| stream_generator.h | 55 bool NextPacket(VCMPacket* packet);
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| rtp_player.h | 84 virtual int NextPacket(int64_t timeNow) = 0;
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