/external/chromium_org/media/cast/net/rtp/ |
H A D | rtp_parser.cc | 50 header->payload_type = bits & ~kRtpMarkerBitMask; 51 if (header->payload_type != expected_payload_type_)
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H A D | rtp_parser_unittest.cc | 33 cast_header_.payload_type = kTestPayloadType; 47 EXPECT_EQ(cast_header_.payload_type, parsed_header.payload_type); 138 cast_header_.payload_type = kTestPayloadType - 1;
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H A D | rtp_packetizer.cc | 16 : payload_type(-1), 128 packet->push_back(static_cast<uint8>(config_.payload_type) |
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H A D | rtp_packetizer_unittest.cc | 48 EXPECT_EQ(kPayload, rtp_header.payload_type); 108 config_.payload_type = kPayload;
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | mediasessionclient_unittest.cc | 1159 buzz::XmlElement* payload_type) = 0; 1161 const buzz::XmlElement* payload_type) = 0; 1163 const buzz::XmlElement* payload_type) = 0; 1165 const buzz::XmlElement* payload_type) = 0; 1213 buzz::XmlElement* NextFromPayloadType(buzz::XmlElement* payload_type) { argument 1214 return payload_type->NextNamed(cricket::QN_JINGLE_RTP_PAYLOADTYPE); 1232 const buzz::XmlElement* payload_type) { 1234 if (payload_type->HasAttr(cricket::QN_ID)) 1235 id = atoi(payload_type->Attr(cricket::QN_ID).c_str()); 1238 if (payload_type 1231 AudioCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1258 VideoCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1289 DataCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1369 NextFromPayloadType(buzz::XmlElement* payload_type) argument 1377 AudioCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1402 VideoCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1427 DataCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1530 NextFromPayloadType(buzz::XmlElement* payload_type) argument 1542 AudioCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1547 VideoCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1552 DataCodecFromPayloadType( const buzz::XmlElement* payload_type) argument 1675 buzz::XmlElement* payload_type = PayloadTypeFromContent(content); local 2227 buzz::XmlElement* payload_type = PayloadTypeFromContent(content); local [all...] |
H A D | mediasessionclient.cc | 758 buzz::XmlElement* payload_type = local 760 AddXmlAttr(payload_type, QN_ID, codec.id); 761 payload_type->AddAttr(QN_NAME, codec.name); 763 AddXmlAttr(payload_type, QN_CLOCKRATE, codec.clockrate); 765 AddXmlAttr(payload_type, QN_BITRATE, codec.bitrate); 767 AddXmlAttr(payload_type, QN_CHANNELS, codec.channels); 768 return payload_type; 772 buzz::XmlElement* payload_type = local 774 AddXmlAttr(payload_type, QN_ID, codec.id); 775 payload_type [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | fec_test_helper.cc | 75 void FrameGenerator::SetRedHeader(Packet* red_packet, uint8_t payload_type, argument 82 red_packet->data[header_length] = payload_type;
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H A D | rtp_rtcp_impl.h | 53 virtual int32_t DeRegisterSendPayload(const int8_t payload_type) OVERRIDE; 103 virtual void SetRtxSendPayloadType(int payload_type) OVERRIDE; 119 const int8_t payload_type, 287 virtual int32_t SetSendREDPayloadType(const int8_t payload_type) OVERRIDE; 290 virtual int32_t SendREDPayloadType(int8_t& payload_type) const OVERRIDE;
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H A D | rtp_rtcp_impl.cc | 250 int* payload_type) const { 251 rtp_sender_.RTXStatus(mode, ssrc, payload_type); 258 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) { argument 259 rtp_sender_.SetRtxPayloadType(payload_type); 309 const int8_t payload_type) { 310 return rtp_sender_.DeRegisterSendPayload(payload_type); 505 int8_t payload_type, 520 payload_type, 555 payload_type, 568 payload_type, 308 DeRegisterSendPayload( const int8_t payload_type) argument 503 SendOutgoingData( FrameType frame_type, int8_t payload_type, uint32_t time_stamp, int64_t capture_time_ms, const uint8_t* payload_data, uint32_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) argument 1026 SetSendREDPayloadType( const int8_t payload_type) argument [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | rtputils.h | 40 int payload_type; member in struct:cricket::RtpHeader
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H A D | rtputils_unittest.cc | 103 EXPECT_EQ(0, header.payload_type); 132 EXPECT_EQ(9, header.payload_type); 148 EXPECT_EQ(9, header.payload_type);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | TestAllCodecs.h | 32 FrameType frame_type, uint8_t payload_type,
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H A D | dual_stream_unittest.cc | 39 FrameType frameType, uint8_t payload_type, 296 int32_t DualStreamTest::SendData(FrameType frameType, uint8_t payload_type, argument 304 if (payload_type == red_encoder_.pltype) { 364 if (payload_type == primary_encoder_.pltype) { 366 } else if (payload_type == secondary_encoder_.pltype) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | decoder_database_unittest.cc | 145 for (uint8_t payload_type = 0; payload_type < kNumPayloads; ++payload_type) { 147 db.RegisterPayload(payload_type, kDecoderArbitrary));
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H A D | payload_splitter_unittest.cc | 107 Packet* CreateOpusFecPacket(uint8_t payload_type, int payload_length, argument 110 packet->header.payloadType = payload_type; 123 Packet* CreatePacket(uint8_t payload_type, int payload_length, argument 126 packet->header.payloadType = payload_type; 139 uint8_t payload_type, 145 EXPECT_EQ(payload_type, packet->header.payloadType); 399 uint8_t payload_type = 0; local 402 VerifyPacket((*it), kPayloadLength, payload_type, kSequenceNumber, 403 kBaseTimestamp, 10 * payload_type); 404 ++payload_type; 137 VerifyPacket(const Packet* packet, int payload_length, uint8_t payload_type, uint16_t sequence_number, uint32_t timestamp, uint8_t payload_value, bool primary = true) argument [all...] |
/external/chromium_org/third_party/webrtc/video/ |
H A D | video_send_stream.cc | 35 ss << ", payload_type: " << payload_type; local 53 ss << ", payload_type: " << payload_type; local 84 if (rtx.payload_type != 0 || !rtx.ssrcs.empty()) 201 assert(config.encoder_settings.payload_type >= 0); 202 assert(config.encoder_settings.payload_type <= 127); 206 config.encoder_settings.payload_type, 262 channel_, config_.encoder_settings.payload_type); 344 video_codec.plType = config_.encoder_settings.payload_type; [all...] |
H A D | video_receive_stream.cc | 67 assert(it->second.payload_type != 0); 70 rtp_rtcp_->SetRtxReceivePayloadType(channel_, it->second.payload_type); 149 decoder.payload_type, 188 channel_, config_.external_decoders[i].payload_type);
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H A D | loopback.cc | 117 send_config.rtp.rtx.payload_type = kRtxPayloadType; 131 send_config.encoder_settings.payload_type = 124; 160 receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_unittest.cc | 50 RtpUtility(int samples_per_packet, uint8_t payload_type) argument 51 : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} 87 uint8_t payload_type, 634 int payload_type, 637 payload_type_ = payload_type; 640 codec_type, channels, payload_type, frame_size_samples); 724 int payload_type, 730 payload_type, 632 RegisterSendCodec(int codec_type, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) argument 722 SetUpTest(int codec_type, int channels, int payload_type, int codec_frame_size_samples, int codec_frame_size_rtp_timestamps) argument
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H A D | audio_coding_module_unittest_oldapi.cc | 49 RtpUtility(int samples_per_packet, uint8_t payload_type) argument 50 : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} 86 uint8_t payload_type, 645 int payload_type, 648 payload_type_ = payload_type; 653 payload_type, 738 int payload_type, 745 payload_type, 642 RegisterSendCodec(const char* payload_name, int sampling_freq_hz, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) argument 735 SetUpTest(const char* codec_name, int codec_sample_rate_hz, int channels, int payload_type, int codec_frame_size_samples, int codec_frame_size_rtp_timestamps) argument
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_rtp_rtcp_impl.h | 43 const uint8_t payload_type); 46 const uint8_t payload_type);
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/external/chromium_org/chrome/renderer/media/ |
H A D | cast_rtp_stream.h | 39 int payload_type; member in struct:CastRtpPayloadParams
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H A D | cast_rtp_stream.cc | 53 payload.payload_type = 127; 69 payload.payload_type = 96; 88 payload.payload_type = 96; 174 config->rtp_payload_type = params.payload.payload_type; 209 config->rtp_payload_type = params.payload.payload_type; 479 : payload_type(0),
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/external/chromium_org/third_party/webrtc/test/ |
H A D | encoder_settings.cc | 61 codec.plType = encoder_settings.payload_type;
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H A D | call_test.cc | 92 send_config_.encoder_settings.payload_type = kFakeSendPayloadType; 109 config.external_decoders[0].payload_type = 110 send_config_.encoder_settings.payload_type;
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