| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| packet_source.h | 34 virtual void FilterOutPayloadType(uint8_t payload_type) {
35 filter_.set(payload_type, true);
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| rtp_generator.cc | 18 uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
28 rtp_header->header.payloadType = payload_type;
47 uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
51 payload_type, payload_length_samples, rtp_header);
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_payload_registry_unittest.cc | 39 RtpUtility::Payload* ExpectReturnOfTypicalAudioPayload(uint8_t payload_type,
53 CreatePayloadType(kTypicalPayloadName, payload_type,
65 uint8_t payload_type = 97; local
67 ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate);
71 kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
77 EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
85 EXPECT_EQ(0, rtp_payload_registry_->DeRegisterReceivePayload(payload_type));
87 payload_type, retrieved_payload));
112 uint8_t payload_type = 97; local
116 ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate)
160 uint8_t payload_type = 97; local
224 int payload_type = GetParam(); local
242 int payload_type = GetParam(); local
[all...] |
| rtp_receiver_audio.cc | 65 int8_t payload_type) const {
67 return (telephone_event_payload_type_ == payload_type) ? true : false;
70 bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type,
77 if (cng_nb_payload_type_ == payload_type) {
84 } else if (cng_wb_payload_type_ == payload_type) {
95 } else if (cng_swb_payload_type_ == payload_type) {
102 } else if (cng_fb_payload_type_ == payload_type) {
110 if (g722_payload_type_ == payload_type) {
119 bool RTPReceiverAudio::ShouldReportCsrcChanges(uint8_t payload_type) const {
121 return !TelephoneEventPayloadType(payload_type);
281 << payload_name << "\/" << payload_type; local
[all...] |
| rtp_receiver_video.h | 44 virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
48 int8_t payload_type,
54 int8_t payload_type,
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| rtp_receiver_audio.h | 43 // Is TelephoneEvent configured with payload type payload_type
44 bool TelephoneEventPayloadType(const int8_t payload_type) const;
50 // Returns true if CNG is configured with payload type payload_type. If so,
52 bool CNGPayloadType(const int8_t payload_type,
69 virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
73 int8_t payload_type,
79 int8_t payload_type,
95 void CheckPayloadChanged(int8_t payload_type,
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| rtp_payload_registry.cc | 40 const int8_t payload_type,
45 assert(payload_type >= 0);
50 switch (payload_type) {
62 << payload_type;
73 payload_type_map_.find(payload_type);
94 LOG(LS_ERROR) << "Payload type already registered: " << payload_type;
107 red_payload_type_ = payload_type;
113 ulpfec_payload_type_ = payload_type;
121 payload_name, payload_type, frequency, channels, rate);
123 payload_type_map_[payload_type] = payload
[all...] |
| rtp_receiver_strategy.h | 61 virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
74 int8_t payload_type,
80 virtual void CheckPayloadChanged(int8_t payload_type,
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| /external/chromium_org/media/cast/net/rtp/ |
| rtp_receiver_defines.cc | 12 payload_type(0),
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| rtp_receiver_defines.h | 20 uint8 payload_type; member in struct:media::cast::RtpCastHeader
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| rtp_header_parser.cc | 30 payload_type(0),
58 const uint8 payload_type = rtp_data_begin_[1] & 0x7f; local
72 parsed_packet->payload_type = payload_type;
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| codec_database.h | 43 uint8_t payload_type,
46 uint8_t payload_type; member in struct:webrtc::VCMExtDecoderMapItem
89 uint8_t payload_type,
95 bool DeregisterExternalEncoder(uint8_t payload_type, bool* was_send_codec);
104 // Deregisters an external decoder object specified by |payload_type|.
105 bool DeregisterExternalDecoder(uint8_t payload_type);
107 // Registers an external decoder object to the payload type |payload_type|.
112 uint8_t payload_type,
121 bool DeregisterReceiveCodec(uint8_t payload_type);
129 // Returns a decoder specified by |payload_type|. The decoded frame callbac
[all...] |
| codec_database.cc | 68 uint8_t payload_type,
70 : payload_type(payload_type),
305 uint8_t payload_type, bool* was_send_codec) {
308 if (external_payload_type_ != payload_type) {
311 if (send_codec_.plType == payload_type) {
326 uint8_t payload_type,
331 external_payload_type_ = payload_type;
427 bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
428 ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| bundlefilter.cc | 54 int payload_type = 0; local
55 if (!GetRtpPayloadType(data, len, &payload_type)) {
58 return FindPayloadType(payload_type);
82 void BundleFilter::AddPayloadType(int payload_type) {
83 payload_types_.insert(payload_type);
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| bundlefilter.h | 57 void AddPayloadType(int payload_type);
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| /external/chromium_org/third_party/webrtc/ |
| video_receive_stream.h | 38 : decoder(NULL), payload_type(0), renderer(false), expected_delay_ms(0) {}
44 int payload_type; member in struct:webrtc::ExternalVideoDecoder
124 Rtx() : ssrc(0), payload_type(0) {}
130 int payload_type; member in struct:webrtc::VideoReceiveStream::Config::Rtp::Rtx
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| video_send_stream.h | 62 EncoderSettings() : payload_type(-1), encoder(NULL) {}
66 int payload_type; member in struct:webrtc::VideoSendStream::Config::EncoderSettings
102 Rtx() : payload_type(-1), pad_with_redundant_payloads(false) {}
108 int payload_type; member in struct:webrtc::VideoSendStream::Config::Rtp::Rtx
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_payload_registry.h | 60 const int8_t payload_type,
67 const int8_t payload_type);
74 int8_t* payload_type) const;
80 void SetRtxPayloadType(int payload_type);
96 bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
98 int GetPayloadTypeFrequency(uint8_t payload_type) const;
100 bool PayloadTypeToPayload(const uint8_t payload_type,
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| rtp_receiver.h | 32 // Is TelephoneEvent configured with payload type payload_type
33 virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
62 const int8_t payload_type,
67 // De-registers |payload_type| from the payload registry.
68 virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_send_test.h | 38 int payload_type,
49 uint8_t payload_type,
|
| acm_send_test_oldapi.h | 40 int payload_type,
51 uint8_t payload_type,
|
| acm_receiver.cc | 477 uint8_t payload_type,
493 if (decoders_[acm_codec_id].payload_type == payload_type &&
502 if (neteq_->RemovePayloadType(decoders_[acm_codec_id].payload_type) !=
505 << decoders_[acm_codec_id].payload_type;
512 ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
515 audio_decoder, neteq_decoder, payload_type);
518 LOG_FERR3(LS_ERROR, "AcmReceiver::AddCodec", acm_codec_id, payload_type,
527 decoders_[acm_codec_id].payload_type = payload_type;
801 uint8_t payload_type = rtp_header.payloadType; local
[all...] |
| acm_receiver.h | 43 uint8_t payload_type; member in struct:webrtc::acm2::AcmReceiver::Decoder
95 // - payload_type : payload type.
113 uint8_t payload_type,
220 // - payload_type : the payload-type to be removed.
225 int RemoveCodec(uint8_t payload_type);
272 // -payload_type : the payload-type of the codec to be retrieved.
281 int DecoderByPayloadType(uint8_t payload_type,
315 int PayloadType2CodecIndex(uint8_t payload_type) const;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 117 int payload_type = -1; local
121 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
124 EXPECT_EQ(kRtxPayloadType, payload_type);
127 payload_type = -1;
129 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
131 EXPECT_EQ(kRtxPayloadType, payload_type);
133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
135 EXPECT_EQ(kRtxPayloadType, payload_type);
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| rtp_player.h | 27 PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
30 payload_type_(payload_type),
35 uint8_t payload_type() const { return payload_type_; } function in class:webrtc::rtpplayer::PayloadCodecTuple
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