| /external/webrtc/media/base/ |
| D | media_engine.cc | 69 const webrtc::RtpParameters& rtp_parameters) { in CheckRtpParametersValues() argument
72 for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { in CheckRtpParametersValues()
73 if (rtp_parameters.encodings[i].bitrate_priority <= 0) { in CheckRtpParametersValues()
78 if (rtp_parameters.encodings[i].scale_resolution_down_by && in CheckRtpParametersValues()
79 *rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) { in CheckRtpParametersValues()
85 if (rtp_parameters.encodings[i].max_framerate && in CheckRtpParametersValues()
86 *rtp_parameters.encodings[i].max_framerate < 0.0) { in CheckRtpParametersValues()
91 if (rtp_parameters.encodings[i].min_bitrate_bps && in CheckRtpParametersValues()
92 rtp_parameters.encodings[i].max_bitrate_bps) { in CheckRtpParametersValues()
93 if (*rtp_parameters.encodings[i].max_bitrate_bps < in CheckRtpParametersValues()
[all …]
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| /external/webrtc/pc/ |
| D | rtp_parameters_conversion_unittest.cc | 616 RtpParameters rtp_parameters = ToRtpParameters<cricket::VideoCodec>( in TEST() local
618 ASSERT_EQ(3u, rtp_parameters.codecs.size()); in TEST()
619 EXPECT_EQ("VP8", rtp_parameters.codecs[0].name); in TEST()
620 EXPECT_EQ("red", rtp_parameters.codecs[1].name); in TEST()
621 EXPECT_EQ("ulpfec", rtp_parameters.codecs[2].name); in TEST()
622 ASSERT_EQ(2u, rtp_parameters.header_extensions.size()); in TEST()
623 EXPECT_EQ("uri", rtp_parameters.header_extensions[0].uri); in TEST()
624 EXPECT_EQ(1, rtp_parameters.header_extensions[0].id); in TEST()
625 EXPECT_EQ("uri2", rtp_parameters.header_extensions[1].uri); in TEST()
626 EXPECT_EQ(3, rtp_parameters.header_extensions[1].id); in TEST()
[all …]
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| D | rtp_parameters_conversion.cc | 436 RtpParameters rtp_parameters; in ToRtpParameters() local
438 rtp_parameters.codecs.push_back(ToRtpCodecParameters(cricket_codec)); in ToRtpParameters()
441 rtp_parameters.header_extensions.emplace_back(cricket_extension.uri, in ToRtpParameters()
444 rtp_parameters.encodings = ToRtpEncodings(stream_params); in ToRtpParameters()
445 return rtp_parameters; in ToRtpParameters()
|
| D | BUILD.gn | 90 "../api:rtp_parameters",
91 "../api:rtp_parameters",
246 "../api:rtp_parameters",
342 "../api:rtp_parameters",
427 "../api:rtp_parameters",
626 "../api:rtp_parameters",
|
| D | rtp_sender.cc | 173 RtpParameters rtp_parameters = parameters; in SetParametersInternal() local
178 rtp_parameters = RestoreEncodingLayers(parameters, disabled_rids_, in SetParametersInternal()
181 return media_channel_->SetRtpSendParameters(ssrc_, rtp_parameters); in SetParametersInternal()
|
| /external/webrtc/test/pc/e2e/media/ |
| D | media_helper.cc | 85 RtpParameters rtp_parameters = sender.value()->GetParameters(); in MaybeAddVideo() local
86 for (auto& encoding_parameters : rtp_parameters.encodings) { in MaybeAddVideo()
90 RTCError res = sender.value()->SetParameters(rtp_parameters); in MaybeAddVideo()
|
| /external/webrtc/api/crypto/ |
| D | BUILD.gn | 36 "..:rtp_parameters",
46 "..:rtp_parameters",
|
| /external/webrtc/api/ |
| D | BUILD.gn | 106 ":rtp_parameters",
173 ":rtp_parameters",
268 ":rtp_parameters",
314 rtc_library("rtp_parameters") {
319 "rtp_parameters.cc",
320 "rtp_parameters.h",
378 ":rtp_parameters",
829 ":rtp_parameters",
846 ":rtp_parameters",
1009 ":rtp_parameters",
|
| /external/webrtc/media/engine/ |
| D | webrtc_video_engine_unittest.cc | 5116 webrtc::RtpParameters rtp_parameters = in TEST_F() local
5118 EXPECT_FALSE(rtp_parameters.rtcp.reduced_size); in TEST_F()
5125 rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); in TEST_F()
5126 EXPECT_TRUE(rtp_parameters.rtcp.reduced_size); in TEST_F()
6868 auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); in TEST_F() local
6869 ASSERT_EQ(3u, rtp_parameters.encodings.size()); in TEST_F()
6870 rtp_parameters.encodings[0].scale_resolution_down_by = 4.0; in TEST_F()
6871 rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; in TEST_F()
6872 rtp_parameters.encodings[2].scale_resolution_down_by = 1.0; in TEST_F()
6873 auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); in TEST_F()
[all …]
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| D | webrtc_voice_engine_unittest.cc | 1299 webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); in TEST_P() local
1300 ASSERT_EQ(2u, rtp_parameters.codecs.size()); in TEST_P()
1301 EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); in TEST_P()
1302 EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); in TEST_P()
1311 webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); in TEST_P() local
1312 EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str()); in TEST_P()
1319 webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); in TEST_P() local
1320 rtp_parameters.header_extensions.emplace_back(); in TEST_P()
1322 EXPECT_NE(0u, rtp_parameters.header_extensions.size()); in TEST_P()
1325 channel_->SetRtpSendParameters(kSsrcX, rtp_parameters); in TEST_P()
[all …]
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| D | webrtc_voice_engine.cc | 974 const webrtc::RtpParameters& rtp_parameters() const { in rtp_parameters() function in cricket::WebRtcVoiceMediaChannel::WebRtcAudioSendStream
1314 webrtc::RtpParameters rtp_parameters; in GetRtpParameters() local
1315 rtp_parameters.encodings.emplace_back(); in GetRtpParameters()
1316 rtp_parameters.encodings[0].ssrc = config_.rtp.remote_ssrc; in GetRtpParameters()
1317 rtp_parameters.header_extensions = config_.rtp.extensions; in GetRtpParameters()
1319 return rtp_parameters; in GetRtpParameters()
1476 webrtc::RtpParameters rtp_params = it->second->rtp_parameters(); in GetRtpSendParameters()
|
| D | webrtc_video_engine.cc | 343 int NumActiveStreams(const webrtc::RtpParameters& rtp_parameters) { in NumActiveStreams() argument
345 for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { in NumActiveStreams()
346 if (rtp_parameters.encodings[i].active) { in NumActiveStreams()
2750 webrtc::RtpParameters rtp_parameters; in GetRtpParameters() local
2755 rtp_parameters.encodings.emplace_back(); in GetRtpParameters()
2756 rtp_parameters.encodings.back().ssrc = ssrc; in GetRtpParameters()
2759 rtp_parameters.header_extensions = config_.rtp.extensions; in GetRtpParameters()
2760 rtp_parameters.rtcp.reduced_size = in GetRtpParameters()
2763 return rtp_parameters; in GetRtpParameters()
|
| /external/webrtc/call/ |
| D | BUILD.gn | 39 "../api:rtp_parameters",
95 "../api:rtp_parameters",
149 "../api:rtp_parameters",
257 "../api:rtp_parameters",
305 "../api:rtp_parameters",
362 "../api:rtp_parameters",
404 "../api:rtp_parameters",
|
| /external/webrtc/logging/ |
| D | BUILD.gn | 43 "../api:rtp_parameters",
189 "../api:rtp_parameters",
309 "../api:rtp_parameters",
364 "../api:rtp_parameters",
|
| /external/webrtc/api/video/ |
| D | BUILD.gn | 260 "..:rtp_parameters",
263 "../:rtp_parameters",
|
| /external/webrtc/video/adaptation/ |
| D | BUILD.gn | 28 "../../api:rtp_parameters",
|
| /external/webrtc/call/adaptation/ |
| D | BUILD.gn | 37 "../../api:rtp_parameters",
|
| /external/webrtc/modules/rtp_rtcp/ |
| D | BUILD.gn | 106 "../../api:rtp_parameters",
260 "../../api:rtp_parameters",
532 "../../api:rtp_parameters",
|
| /external/webrtc/media/ |
| D | BUILD.gn | 92 "../api:rtp_parameters",
288 "../api:rtp_parameters",
554 "../api:rtp_parameters",
|
| /external/webrtc/video/ |
| D | BUILD.gn | 75 "../api:rtp_parameters",
217 "../api:rtp_parameters",
580 "../api:rtp_parameters",
|
| /external/webrtc/test/scenario/ |
| D | BUILD.gn | 84 "../../api:rtp_parameters",
|
| /external/webrtc/audio/ |
| D | BUILD.gn | 48 "../api:rtp_parameters",
|
| /external/webrtc/test/ |
| D | BUILD.gn | 199 "../api:rtp_parameters",
839 "../api:rtp_parameters",
|
| /external/webrtc/sdk/android/ |
| D | BUILD.gn | 698 "src/jni/pc/rtp_parameters.cc",
699 "src/jni/pc/rtp_parameters.h",
730 "../../api:rtp_parameters",
|
| /external/webrtc/test/pc/e2e/ |
| D | BUILD.gn | 714 "../../../api:rtp_parameters",
|