/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module.cc | 45 int AudioCodingModule::Codec(const char* payload_name, argument 53 payload_name, sampling_freq_hz, channels); 76 int AudioCodingModule::Codec(const char* payload_name, argument 79 return acm2::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
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H A D | acm_send_test_oldapi.h | 37 bool RegisterCodec(const char* payload_name,
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H A D | acm_send_test_oldapi.cc | 48 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name, argument 55 payload_name, &codec_, sampling_freq_hz, channels));
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H A D | acm_codec_database.cc | 510 int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) { argument 519 name_match = (STR_CASE_CMP(database_[id].plname, payload_name) == 0); 522 if (STR_CASE_CMP(payload_name, "opus") != 0) {
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H A D | acm_codec_database.h | 267 static int CodecId(const char* payload_name, int frequency, int channels);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_receiver_video.h | 47 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 55 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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H A D | rtp_receiver_audio.h | 72 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 80 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 87 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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H A D | rtp_payload_registry.cc | 39 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 46 assert(payload_name); 68 size_t payload_name_length = strlen(payload_name); 87 payload->name, payload_name, payload_name_length)) { 100 payload_name, payload_name_length, frequency, channels, rate); 106 if (RtpUtility::StringCompare(payload_name, "red", 3)) { 111 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); 112 } else if (RtpUtility::StringCompare(payload_name, "ulpfec", 3)) { 117 strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1); 121 payload_name, payload_typ 38 RegisterReceivePayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_type, const uint32_t frequency, const uint8_t channels, const uint32_t rate, bool* created_new_payload) argument 146 DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const size_t payload_name_length, const uint32_t frequency, const uint8_t channels, const uint32_t rate) argument 179 ReceivePayloadType( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const uint32_t frequency, const uint8_t channels, const uint32_t rate, int8_t* payload_type) const argument [all...] |
H A D | rtp_receiver_impl.cc | 99 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 111 payload_name, payload_type, frequency, channels, rate, 114 if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type, 116 LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/" 268 char payload_name[RTP_PAYLOAD_NAME_SIZE]; local 300 payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; 301 strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1); 320 id_, rtp_header.payloadType, payload_name, 344 char payload_name[RTP_PAYLOAD_NAME_SIZE]; local 391 payload_name[RTP_PAYLOAD_NAME_SIZ 98 RegisterReceivePayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_type, const uint32_t frequency, const uint8_t channels, const uint32_t rate) argument [all...] |
H A D | rtp_receiver_video.cc | 44 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 97 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 102 id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) { 43 OnNewPayloadTypeCreated( const char payload_name[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency) argument 93 InvokeOnInitializeDecoder( RtpFeedback* callback, int32_t id, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion& specific_payload) const argument
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H A D | rtp_receiver_audio.cc | 157 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 162 if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) { 165 if (RtpUtility::StringCompare(payload_name, "cn", 2)) { 272 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 276 payload_name, 281 << payload_name << "/" << payload_type; 156 OnNewPayloadTypeCreated( const char payload_name[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency) argument 268 InvokeOnInitializeDecoder( RtpFeedback* callback, int32_t id, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion& specific_payload) const argument
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H A D | rtp_receiver_strategy.h | 75 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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H A D | rtp_receiver_impl.h | 38 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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H A D | rtp_sender_unittest.cc | 739 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local 741 ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 826 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local 828 ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 881 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local 885 rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 1500)); 965 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC"; local 967 ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 1014 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; local 1016 ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_typ 1043 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME"; local 1083 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event"; local [all...] |
H A D | rtp_sender.cc | 224 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 227 assert(payload_name); 240 payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) { 258 ret_val = audio_->RegisterAudioPayload(payload_name, payload_number, 261 ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate, 223 RegisterPayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], const int8_t payload_number, const uint32_t frequency, const uint8_t channels, const uint32_t rate) argument
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/external/chromium_org/third_party/webrtc/test/ |
H A D | encoder_settings.cc | 62 strcpy(codec.plName, encoder_settings.payload_name.c_str()); 63 if (encoder_settings.payload_name == "VP8") { 65 } else if (encoder_settings.payload_name == "H264") {
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H A D | call_test.cc | 91 send_config_.encoder_settings.payload_name = "FAKE";
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_payload_registry.h | 59 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 70 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 142 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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H A D | rtp_receiver.h | 61 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | codec.cc | 177 const char* payload_name = name.c_str(); local 178 if (_stricmp(payload_name, kRedCodecName) == 0) { 181 if (_stricmp(payload_name, kUlpfecCodecName) == 0) { 184 if (_stricmp(payload_name, kRtxCodecName) == 0) {
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/external/chromium_org/third_party/webrtc/ |
H A D | video_send_stream.h | 65 std::string payload_name; member in struct:webrtc::VideoSendStream::Config::EncoderSettings
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/interface/ |
H A D | audio_coding_module.h | 141 // -payload_name : name of the codec. 154 static int Codec(const char* payload_name, CodecInst* codec, 164 // -payload_name : name of the codec. 173 static int Codec(const char* payload_name, int sampling_freq_hz,
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/external/chromium_org/third_party/webrtc/video/ |
H A D | video_send_stream.cc | 34 ss << "{payload_name: " << payload_name; local 306 if (config_.encoder_settings.payload_name == "VP8") { 308 } else if (config_.encoder_settings.payload_name == "H264") { 341 config_.encoder_settings.payload_name.c_str(),
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H A D | loopback.cc | 130 send_config.encoder_settings.payload_name = flags::Codec();
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H A D | replay.cc | 213 encoder_settings.payload_name = flags::Codec();
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