/external/webrtc/logging/rtc_event_log/ |
D | rtc_stream_config.cc | 33 StreamConfig::Codec::Codec(const std::string& payload_name, in Codec() argument 36 : payload_name(payload_name), in Codec() 41 return payload_name == other.payload_name && in operator ==()
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D | rtc_stream_config.h | 44 Codec(const std::string& payload_name, 50 std::string payload_name; member
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/external/webrtc/video/end_to_end_tests/ |
D | multi_codec_receive_tests.cc | 37 uint8_t PayloadNameToPayloadType(const std::string& payload_name) { in PayloadNameToPayloadType() argument 38 if (payload_name == "VP8") { in PayloadNameToPayloadType() 40 } else if (payload_name == "VP9") { in PayloadNameToPayloadType() 42 } else if (payload_name == "H264") { in PayloadNameToPayloadType() 162 std::string payload_name; member 186 if (unique_payload_names.insert(config.payload_name).second) { in ConfigureDecoders() 188 PayloadNameToPayloadType(config.payload_name), config.payload_name); in ConfigureDecoders() 198 GetVideoSendConfig()->rtp.payload_name = config.payload_name; in ConfigureEncoder() 200 PayloadNameToPayloadType(config.payload_name); in ConfigureEncoder() 202 PayloadStringToCodecType(config.payload_name); in ConfigureEncoder() [all …]
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D | codec_tests.cc | 53 const std::string& payload_name, in CodecObserver() argument 62 payload_name_(payload_name), in CodecObserver() 78 send_config->rtp.payload_name = payload_name_; in ModifyVideoConfigs() 86 SdpVideoFormat(send_config->rtp.payload_name); in ModifyVideoConfigs()
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D | extended_reports_tests.cc | 180 send_config->rtp.payload_name = "VP8"; in ModifyVideoConfigs() 186 SdpVideoFormat(send_config->rtp.payload_name); in ModifyVideoConfigs()
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D | frame_encryption_tests.cc | 42 send_config->rtp.payload_name = "VP8"; in ModifyVideoConfigs()
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D | fec_tests.cc | 118 send_config->rtp.payload_name = "VP8"; in TEST_F() 477 send_config->rtp.payload_name = "VP8"; in TEST_F() 489 SdpVideoFormat(send_config->rtp.payload_name); in TEST_F()
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D | multi_stream_tester.cc | 90 send_config.rtp.payload_name = "VP8"; in RunTest()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_audio_unittest.cc | 90 const char payload_name[] = "PAYLOAD_NAME"; in TEST_F() local 93 payload_name, payload_type, 48000, 0, 1500)); in TEST_F() 110 const char payload_name[] = "PAYLOAD_NAME"; in TEST_F() local 113 payload_name, payload_type, 48000, 0, 1500)); in TEST_F() 135 const char payload_name[] = "audio"; in TEST_F() local 138 payload_name, payload_type, 48000, 0, 1500)); in TEST_F() 153 const char payload_name[] = "audio"; in TEST_F() local 156 payload_name, payload_type, 48000, 0, 1500)); in TEST_F()
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D | rtp_sender_audio.cc | 63 int32_t RTPSenderAudio::RegisterAudioPayload(absl::string_view payload_name, in RegisterAudioPayload() argument 68 if (absl::EqualsIgnoreCase(payload_name, "cn")) { in RegisterAudioPayload() 87 } else if (absl::EqualsIgnoreCase(payload_name, "telephone-event")) { in RegisterAudioPayload() 94 } else if (payload_name == "audio") { in RegisterAudioPayload()
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D | rtp_sender_audio.h | 37 int32_t RegisterAudioPayload(absl::string_view payload_name,
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/external/webrtc/rtc_tools/rtp_generator/ |
D | rtp_generator.cc | 98 &config.rtp.payload_name)) { in ParseVideoSendStreamConfig() 102 if (!IsValidCodecType(config.rtp.payload_name)) { in ParseVideoSendStreamConfig() 108 GetDefaultTypeForPayloadName(config.rtp.payload_name); in ParseVideoSendStreamConfig() 188 PayloadStringToCodecType(video_config.rtp.payload_name); in RtpGenerator() 189 if (video_config.rtp.payload_name == cricket::kVp8CodecName) { in RtpGenerator() 193 } else if (video_config.rtp.payload_name == cricket::kVp9CodecName) { in RtpGenerator() 197 } else if (video_config.rtp.payload_name == cricket::kH264CodecName) { in RtpGenerator() 202 encoder_config.video_format.name = video_config.rtp.payload_name; in RtpGenerator() 221 video_config.rtp.payload_name, /*max qp*/ 56, /*screencast*/ false, in RtpGenerator()
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/external/webrtc/video/ |
D | quality_scaling_tests.cc | 58 const std::string& payload_name, 69 const std::string& payload_name, in RunTest() argument 79 const std::string& payload_name, in RunTest() argument 86 payload_name_(payload_name), in RunTest() 116 send_config->rtp.payload_name = payload_name_; in RunTest() 136 } test(encoder_factory, payload_name, start_bps, automatic_resize, in RunTest()
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D | picture_id_tests.cc | 224 const std::string& payload_name); 241 const std::string& payload_name) { in SetupEncoder() argument 243 new PictureIdObserver(PayloadStringToCodecType(payload_name))); in SetupEncoder() 246 RTC_FROM_HERE, task_queue(), [this, encoder_factory, payload_name]() { in SetupEncoder() 260 GetVideoSendConfig()->rtp.payload_name = payload_name; in SetupEncoder() 262 PayloadStringToCodecType(payload_name); in SetupEncoder()
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D | send_statistics_proxy.cc | 68 const std::string& payload_name) { in PayloadNameToHistogramCodecType() argument 69 VideoCodecType codecType = PayloadStringToCodecType(payload_name); in PayloadNameToHistogramCodecType() 82 void UpdateCodecTypeHistogram(const std::string& payload_name) { in UpdateCodecTypeHistogram() argument 84 PayloadNameToHistogramCodecType(payload_name), in UpdateCodecTypeHistogram() 137 payload_name_(config.rtp.payload_name), in SendStatisticsProxy()
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/external/webrtc/modules/audio_coding/acm2/ |
D | acm_send_test.cc | 61 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name, in RegisterCodec() argument 66 SdpAudioFormat format(payload_name, clockrate_hz, num_channels); in RegisterCodec() 67 if (absl::EqualsIgnoreCase(payload_name, "g722")) { in RegisterCodec() 70 } else if (absl::EqualsIgnoreCase(payload_name, "opus")) { in RegisterCodec()
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D | acm_send_test.h | 39 bool RegisterCodec(const char* payload_name,
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/external/webrtc/test/ |
D | encoder_settings.cc | 151 const std::string& payload_name) { in CreateMatchingDecoder() argument 154 decoder.video_format = SdpVideoFormat(payload_name); in CreateMatchingDecoder() 161 config.rtp.payload_name); in CreateMatchingDecoder()
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D | encoder_settings.h | 58 const std::string& payload_name);
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/external/autotest/client/site_tests/logging_CrashSender/ |
D | logging_CrashSender.py | 36 def _check_send_result(self, result, report_kind, payload_name, argument 54 desired_payload = self.get_crash_dir_name(payload_name)
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/external/webrtc/media/engine/ |
D | fake_webrtc_call.cc | 234 if (config_.rtp.payload_name == "VP8") { in ReconfigureVideoEncoder() 241 } else if (config_.rtp.payload_name == "VP9") { in ReconfigureVideoEncoder() 248 } else if (config_.rtp.payload_name == "H264") { in ReconfigureVideoEncoder() 255 << config_.rtp.payload_name; in ReconfigureVideoEncoder()
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/external/webrtc/call/ |
D | rtp_config.h | 110 std::string payload_name; member
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D | bitrate_estimator_tests.cc | 132 video_send_config.rtp.payload_name = "FAKE"; in SetUp() 197 SdpVideoFormat(test_->GetVideoSendConfig()->rtp.payload_name);
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D | rtp_video_sender.cc | 58 bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name, in PayloadTypeSupportsSkippingFecPackets() argument 60 const VideoCodecType codecType = PayloadStringToCodecType(payload_name); in PayloadTypeSupportsSkippingFecPackets() 105 !PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name, trials)) { in ShouldDisableRedAndUlpfec() 310 return PayloadStringToCodecType(config.payload_name); in GetVideoCodecType()
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/external/webrtc/logging/rtc_event_log/encoder/ |
D | rtc_event_log_encoder_legacy.cc | 631 decoder->set_name(d.payload_name); in EncodeVideoReceiveStreamConfig() 671 encoder->set_name(codec.payload_name); in EncodeVideoSendStreamConfig() 678 << codec.payload_name; in EncodeVideoSendStreamConfig()
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