Reintroduce division by two for audio playout delay
When migrating the audio device, we accidentally dropped a /2 for PlayoutDelay. This meant we would estimate a delay of 150ms instead of 75ms for JavaAudioDeviceModules. This change fixes that. Bug: webrtc:7452 Change-Id: I20b70ebf141410209953243ae665644b92e480f5 Reviewed-on: https://webrtc-review.googlesource.com/c/113946 Commit-Queue: Paulina Hensman <phensman@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25986}
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@ -747,12 +747,12 @@ TEST_F(AudioDeviceTest,
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// selected audio layer. Note that, this delay estimate will only be utilized
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// if the HW AEC is disabled.
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TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
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EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds,
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EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds / 2,
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TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio));
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}
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TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
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EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
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EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds / 2,
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TestDelayOnAudioLayer(
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AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
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}
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@ -505,7 +505,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
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int32_t PlayoutDelay(uint16_t* delay_ms) const override {
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// Best guess we can do is to use half of the estimated total delay.
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*delay_ms = playout_delay_ms_;
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*delay_ms = playout_delay_ms_ / 2;
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RTC_DCHECK_GT(*delay_ms, 0);
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return 0;
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}
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