Searched refs:num_audio_streams_ (Results 1 – 5 of 5) sorted by relevance
| /aosp_15_r20/external/webrtc/call/ |
| H A D | rampup_tests.cc | 76 num_audio_streams_(num_audio_streams), in RampUpTester()
93 audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)), in RampUpTester()
97 EXPECT_LE(num_audio_streams_, 1u); in RampUpTester()
135 return num_audio_streams_; in GetNumAudioStreams()
266 if (num_audio_streams_ == 0) in ModifyAudioConfigs()
475 if (num_audio_streams_ > 0 && sender_call_) { in PollStats()
497 if (num_audio_streams_ > 0) { in GetModifierString()
498 str += rtc::ToString(num_audio_streams_); in GetModifierString()
500 str += (num_audio_streams_ > 1 ? "s" : ""); in GetModifierString()
512 if (num_audio_streams_ > 0) in GetExpectedHighBitrate()
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| H A D | rampup_tests.h | 78 const size_t num_audio_streams_; variable
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| /aosp_15_r20/external/webrtc/test/ |
| H A D | call_test.cc | 56 num_audio_streams_(0),
92 num_audio_streams_ = test->GetNumAudioStreams(); in RunBaseTest()
94 RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); in RunBaseTest()
97 if (num_audio_streams_ > 0) { in RunBaseTest()
117 if (num_audio_streams_ > 0) { in RunBaseTest()
138 if (num_audio_streams_ > 0) in RunBaseTest()
146 CreateSendConfig(num_video_streams_, num_audio_streams_, in RunBaseTest()
155 if (num_audio_streams_ > 0) { in RunBaseTest()
170 if (num_audio_streams_ > 0) { in RunBaseTest()
403 RTC_DCHECK_GE(1, num_audio_streams_); in CreateMatchingAudioAndFecConfigs()
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| H A D | call_test.h | 225 size_t num_audio_streams_; variable
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| /aosp_15_r20/external/webrtc/video/end_to_end_tests/ |
| H A D | transport_feedback_tests.cc | 253 num_audio_streams_(num_audio_streams), in TransportFeedbackTester()
291 size_t GetNumAudioStreams() const override { return num_audio_streams_; } in GetNumAudioStreams()
315 const size_t num_audio_streams_; member in webrtc::TransportFeedbackTester
355 num_audio_streams_(num_audio_streams), in TEST_F()
423 size_t GetNumAudioStreams() const override { return num_audio_streams_; } in TEST_F()
427 const size_t num_audio_streams_; in TEST_F() member in webrtc::TEST_F::TransportFeedbackTester
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