From d8bbc7858622b6d9c278469aab701ca0b609cddf Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 15 May 2024 05:35:49 +0200 Subject: Merging upstream version 126.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc') diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc index 86240da196..f483b8dc79 100644 --- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc +++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc @@ -13,6 +13,7 @@ #include #include +#include "api/array_view.h" #include "modules/audio_device/audio_device_buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" @@ -107,7 +108,8 @@ void FineAudioBuffer::GetPlayoutData(rtc::ArrayView audio_buffer, void FineAudioBuffer::DeliverRecordedData( rtc::ArrayView audio_buffer, - int record_delay_ms) { + int record_delay_ms, + absl::optional capture_time_ns) { RTC_DCHECK(IsReadyForRecord()); // Always append new data and grow the buffer when needed. record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size()); @@ -118,7 +120,8 @@ void FineAudioBuffer::DeliverRecordedData( record_channels_ * record_samples_per_channel_10ms_; while (record_buffer_.size() >= num_elements_10ms) { audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(), - record_samples_per_channel_10ms_); + record_samples_per_channel_10ms_, + capture_time_ns); audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms); audio_device_buffer_->DeliverRecordedData(); memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms, -- cgit v1.2.3