| /external/chromium_org/media/ffmpeg/ |
| ffmpeg_common.h | 58 AVPacket* packet = static_cast<AVPacket*>(x); local
59 av_free_packet(packet);
60 delete packet;
|
| /external/android-clat/ |
| ipv4.c | 27 * translates an icmp packet
28 * out - output packet
29 * icmp - pointer to icmp header in packet
51 * translates an ipv4 packet
52 * out - output packet
53 * packet - packet data
54 * len - size of packet
57 int ipv4_packet(clat_packet out, clat_packet_index pos, const uint8_t *packet, size_t len) {
58 const struct iphdr *header = (struct iphdr *) packet;
[all...] |
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host.cc | 33 // enabled. HMAC in packet will be compared against this value before updating
34 // packet with actual HMAC value.
160 // Assumes |length| is actual packet length + tag length. Updates HMAC at end of
161 // the RTP packet.
197 // Copy ROC after end of rtp packet.
199 // Authentication of a RTP packet will have RTP packet + ROC size.
208 // Copy HMAC from output to packet. This is required as auth tag length
226 // PacketOptions, nothing to be updated in this packet.
235 // If there is a srtp auth key present then packet must be a RTP packet
[all...] |
| socket_host_test_utils.h | 87 void CreateRandomPacket(std::vector<char>* packet);
88 void CreateStunRequest(std::vector<char>* packet);
89 void CreateStunResponse(std::vector<char>* packet);
90 void CreateStunError(std::vector<char>* packet);
|
| /external/chromium_org/media/cast/test/ |
| loopback_transport.h | 25 // Class that sends the packet to a receiver through a stack of PacketPipes.
32 virtual bool SendPacket(PacketRef packet,
|
| /external/chromium_org/media/filters/ |
| audio_decoder_unittest.cc | 66 static void SetDiscardPadding(AVPacket* packet,
85 packet, AV_PKT_DATA_SKIP_SAMPLES, &skip_samples_size));
141 // Load the first packet and check its timestamp.
142 AVPacket packet; local
143 ASSERT_TRUE(reader_->ReadPacketForTesting(&packet));
144 EXPECT_EQ(GetParam().first_packet_pts, packet.pts);
146 reader_->GetAVStreamForTesting()->time_base, packet.pts);
147 av_free_packet(&packet);
177 AVPacket packet; local
178 ASSERT_TRUE(reader_->ReadPacketForTesting(&packet));
[all...] |
| ffmpeg_demuxer.cc | 177 void FFmpegDemuxerStream::EnqueuePacket(ScopedAVPacket packet) {
181 NOTREACHED() << "Attempted to enqueue packet on a stopped stream";
186 // Convert the packet if there is a bitstream filter.
187 if (packet->data && bitstream_converter_enabled_ &&
188 !bitstream_converter_->ConvertPacket(packet.get())) {
196 av_packet_split_side_data(packet.get());
203 packet.get(),
209 packet.get(),
218 buffer = DecoderBuffer::CopyFrom(packet.get()->data, packet.get()->size
[all...] |
| /external/chromium_org/media/midi/ |
| usb_midi_input_stream.cc | 69 const uint8* packet,
71 // The first 4 bytes of the packet is accessible here.
72 uint8 code_index = packet[0] & 0x0f;
73 uint8 cable_number = packet[0] >> 4;
89 delegate_->OnReceivedData(it->second, &packet[1], packet_size, time);
|
| /external/chromium_org/net/quic/ |
| quic_connection_logger.h | 40 const QuicEncryptedPacket& packet,
47 const QuicEncryptedPacket& packet) OVERRIDE;
68 const QuicPublicResetPacket& packet) OVERRIDE;
70 const QuicVersionNegotiationPacket& packet) OVERRIDE;
90 // packet sequence numbers, that was gathered in the vectors
95 // Do a factory get for a histogram to record a 6-packet loss-sequence as a
101 // packet sequence. |which_21| is used to adjust the name of the histogram
102 // to distinguish the first 21 packets' loss data, vs. some later 21 packet
107 // corresponds to the oldest packet sequence number in the series of packets,
109 // packet. Of the maximum of 21 bits that are valid (correspond to packets)
[all...] |
| /external/chromium_org/remoting/client/ |
| audio_decode_scheduler.h | 38 virtual void ProcessAudioPacket(scoped_ptr<AudioPacket> packet,
|
| audio_player.h | 20 void ProcessAudioPacket(scoped_ptr<AudioPacket> packet);
|
| /external/chromium_org/remoting/codec/ |
| video_decoder_verbatim.h | 16 // Video decoder implementations that decodes video packet encoded by
27 virtual bool DecodePacket(const VideoPacket& packet) OVERRIDE;
|
| /external/chromium_org/remoting/protocol/ |
| audio_reader.h | 39 void OnNewData(scoped_ptr<AudioPacket> packet,
|
| monitored_video_stub.h | 25 // video channel is connected and forward the packet to the underlying
43 virtual void ProcessVideoPacket(scoped_ptr<VideoPacket> packet,
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| testutils.cc | 75 const RawRtpPacket& packet, uint16 seq, uint32 ts, uint32 ssc) const {
78 ver_to_cc == packet.ver_to_cc &&
79 m_to_pt == packet.m_to_pt &&
81 0 == memcmp(payload, packet.payload, sizeof(payload));
107 bool RawRtcpPacket::EqualsTo(const RawRtcpPacket& packet) const {
108 return ver_to_count == packet.ver_to_count &&
109 type == packet.type &&
110 length == packet.length &&
111 0 == memcmp(payload, packet.payload, sizeof(payload));
171 RtpDumpPacket packet; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| mediarecorder_unittest.cc | 75 rtc::StreamResult ReadPacket(RtpDumpPacket* packet) {
81 return reader_->ReadPacket(packet);
92 // By default, the sink is disabled. The 1st packet is not written.
97 // Enable the sink. The 2nd packet is written.
103 // Disable the sink. The 3rd packet is not written.
108 // Read the recorded file and verify it contains only the 2nd packet.
109 RtpDumpPacket packet; local
110 EXPECT_EQ(rtc::SR_SUCCESS, ReadPacket(&packet));
112 &packet, &RtpTestUtility::kTestRawRtpPackets[1], false));
113 EXPECT_EQ(rtc::SR_EOS, ReadPacket(&packet));
130 RtpDumpPacket packet; local
160 RtpDumpPacket packet; local
[all...] |
| /external/chromium_org/third_party/mesa/src/src/mesa/drivers/dri/r200/ |
| r200_ioctl.h | 147 #define OUT_BATCH_PACKET3(packet, num_extra) do { \
149 OUT_BATCH(CP_PACKET3((packet), (num_extra))); \
152 #define OUT_BATCH_PACKET3_CLIP(packet, num_extra) do { \
154 OUT_BATCH(CP_PACKET3((packet), (num_extra))); \
|
| /external/chromium_org/third_party/mesa/src/src/mesa/drivers/dri/radeon/ |
| radeon_ioctl.h | 160 #define OUT_BATCH_PACKET3(packet, num_extra) do { \
162 OUT_BATCH(CP_PACKET3((packet), (num_extra))); \
165 #define OUT_BATCH_PACKET3_CLIP(packet, num_extra) do { \
167 OUT_BATCH(CP_PACKET3((packet), (num_extra))); \
|
| /external/chromium_org/third_party/speex/include/speex/ |
| speex_header.h | 71 spx_int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */
80 /** Creates the header packet from the header itself (mostly involves endianness conversion) */
83 /** Creates a SpeexHeader from a packet */
84 SpeexHeader *speex_packet_to_header(char *packet, int size);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_send_test.cc | 21 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
62 Packet* AcmSendTest::NextPacket() {
66 // same throughout the whole test run, no packet at all will be delivered.
70 // Insert audio and process until one packet is produced.
84 // Encoded packet received.
92 // This method receives the callback from ACM when a new packet is produced.
99 // Store the packet locally.
108 Packet* AcmSendTest::CreatePacket() {
133 Packet* packet local
[all...] |
| acm_send_test_oldapi.cc | 21 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
65 Packet* AcmSendTestOldApi::NextPacket() {
69 // same throughout the whole test run, no packet at all will be delivered.
73 // Insert audio and process until one packet is produced.
87 // Encoded packet received.
95 // This method receives the callback from ACM when a new packet is produced.
103 // Store the packet locally.
112 Packet* AcmSendTestOldApi::CreatePacket() {
137 Packet* packet local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| comfort_noise.h | 23 struct Packet;
48 // Update the comfort noise generator with the parameters in |packet|.
49 // Will delete the packet.
50 int UpdateParameters(Packet* packet);
|
| timestamp_scaler.h | 15 #include "webrtc/modules/audio_coding/neteq/packet.h"
41 // Scale the timestamp in |packet| from external to internal.
42 virtual void ToInternal(Packet* packet);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_file_source.cc | 21 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
49 Packet* RtpFileSource::NextPacket() {
76 // May be an RTCP packet.
86 scoped_ptr<Packet> packet(new Packet(packet_memory.release(),
91 if (!packet->valid_header()) {
95 if (filter_.test(packet->header().payloadType)) {
96 // This payload type should be filtered out. Continue to the next packet.
99 return packet.release()
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| replay.cc | 239 test::RtpFileReader::Packet packet; local
240 if (!rtp_reader->NextPacket(&packet))
243 switch (call->Receiver()->DeliverPacket(packet.data, packet.length)) {
249 parser->Parse(packet.data, packet.length, &header);
256 fprintf(stderr, "Packet error, corrupt packets or incorrect setup?\n");
259 if (last_time_ms != 0 && last_time_ms != packet.time_ms) {
260 SleepMs(packet.time_ms - last_time_ms)
[all...] |
