| /external/webrtc/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 | 27 // NextPacket method.
30 // * packet_size_in_bytes is the number of bytes to read in each NextPacket
41 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/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet_source.h | 31 virtual Packet* NextPacket() = 0;
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| constant_pcm_packet_source.h | 36 Packet* NextPacket() override;
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| rtc_event_log_source.h | 46 Packet* NextPacket() override;
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| rtp_file_source.h | 48 Packet* NextPacket() override;
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| rtpcat.cc | 41 while (input->NextPacket(&packet))
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| rtp_file_source.cc | 56 Packet* RtpFileSource::NextPacket() {
59 if (!rtp_reader_->NextPacket(&temp_packet)) {
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| constant_pcm_packet_source.cc | 38 Packet* ConstantPcmPacketSource::NextPacket() {
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| fec_test_helper.h | 37 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 | 44 bool NextPacket(uint8_t* buffer,
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| fec_test_helper.cc | 28 RtpPacket* FrameGenerator::NextPacket(int offset, size_t length) {
67 RtpPacket* red_packet = NextPacket(0, packet->length + 1);
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| rtp_format_h264.h | 40 bool NextPacket(uint8_t* buffer,
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| rtp_format_vp9.h | 16 // After creating the packetizer, the method NextPacket is called
58 bool NextPacket(uint8_t* buffer,
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| /external/webrtc/webrtc/test/ |
| rtp_file_reader.h | 45 virtual bool NextPacket(RtpPacket* packet) = 0;
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| rtp_file_reader_unittest.cc | 35 while (rtp_packet_source_->NextPacket(&packet)) {
75 while (rtp_packet_source_->NextPacket(&packet)) {
85 while (rtp_packet_source_->NextPacket(&packet)) {
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| packet_buffer_unittest.cc | 31 Packet* NextPacket(int payload_size_bytes);
52 Packet* PacketGenerator::NextPacket(int payload_size_bytes) {
93 Packet* packet = gen.NextPacket(payload_len);
116 Packet* packet = gen.NextPacket(payload_len);
137 Packet* packet = gen.NextPacket(payload_len);
146 Packet* packet = gen.NextPacket(payload_len);
166 Packet* packet = gen.NextPacket(payload_len);
202 Packet* packet = gen.NextPacket(payload_len);
206 Packet* packet = gen.NextPacket(payload_len);
270 Packet* packet = gen.NextPacket(kPayloadLength)
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_send_test_oldapi.h | 50 Packet* NextPacket();
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| stream_generator.h | 46 bool NextPacket(VCMPacket* packet);
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| /external/webrtc/webrtc/base/ |
| testclient.cc | 19 // NextPacket.
54 TestClient::Packet* TestClient::NextPacket(int timeout_ms) {
91 Packet* packet = NextPacket(kTimeoutMs);
103 Packet* packet = NextPacket(kNoPacketTimeoutMs);
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| testclient.h | 35 // Default timeout for NextPacket reads.
63 Packet* NextPacket(int timeout_ms);
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
| bwe_rtp_play.cc | 62 if (!rtp_reader->NextPacket(&packet)) {
88 if (!rtp_reader->NextPacket(&packet)) {
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| rtp_to_text.cc | 37 while (rtp_reader->NextPacket(&packet)) {
|
