webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h - Issue 1853183002: Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

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Unified Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

Issue 1853183002: Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t> (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Adding back the old PlayoutTimestamp method, now DEPRECATED Created 4 years, 8 months ago

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Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

index e463d29f9b53dddf3e94d5804371d2594b8d5583..63dfb8105670824c9b4452be6ab15aa6ef80702b 100644

--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

@@ -157,8 +157,9 @@ class AudioCodingModuleImpl final : public AudioCodingModule {

// Smallest latency NetEq will maintain.

int LeastRequiredDelayMs() const override;

- // Get playout timestamp.

- int PlayoutTimestamp(uint32_t* timestamp) override;

+ RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;

+ rtc::Optional<uint32_t> PlayoutTimestamp() override;

// Get 10 milliseconds of raw audio data to play out, and

// automatic resample to the requested frequency if > 0.