diff options
Diffstat (limited to '')
| -rw-r--r-- | third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc | 7 |
1 files changed, 5 insertions, 2 deletions
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 <cstdint> #include <cstring> +#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<int16_t> audio_buffer, void FineAudioBuffer::DeliverRecordedData( rtc::ArrayView<const int16_t> audio_buffer, - int record_delay_ms) { + int record_delay_ms, + absl::optional<int64_t> 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, |