1 files changed, 130 insertions, 0 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
new file mode 100644
index 0000000000..86240da196
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/fine_audio_buffer.h"
+
+#include <cstdint>
+#include <cstring>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer)
+    : audio_device_buffer_(audio_device_buffer),
+      playout_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->PlayoutSampleRate() * 10 / 1000)),
+      record_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->RecordingSampleRate() * 10 / 1000)),
+      playout_channels_(audio_device_buffer->PlayoutChannels()),
+      record_channels_(audio_device_buffer->RecordingChannels()) {
+  RTC_DCHECK(audio_device_buffer_);
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (IsReadyForPlayout()) {
+    RTC_DLOG(LS_INFO) << "playout_samples_per_channel_10ms: "
+                      << playout_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "playout_channels: " << playout_channels_;
+  }
+  if (IsReadyForRecord()) {
+    RTC_DLOG(LS_INFO) << "record_samples_per_channel_10ms: "
+                      << record_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "record_channels: " << record_channels_;
+  }
+}
+
+FineAudioBuffer::~FineAudioBuffer() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+void FineAudioBuffer::ResetPlayout() {
+  playout_buffer_.Clear();
+}
+
+void FineAudioBuffer::ResetRecord() {
+  record_buffer_.Clear();
+}
+
+bool FineAudioBuffer::IsReadyForPlayout() const {
+  return playout_samples_per_channel_10ms_ > 0 && playout_channels_ > 0;
+}
+
+bool FineAudioBuffer::IsReadyForRecord() const {
+  return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
+}
+
+void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
+                                     int playout_delay_ms) {
+  RTC_DCHECK(IsReadyForPlayout());
+  // Ask WebRTC for new data in chunks of 10ms until we have enough to
+  // fulfill the request. It is possible that the buffer already contains
+  // enough samples from the last round.
+  while (playout_buffer_.size() < audio_buffer.size()) {
+    // Get 10ms decoded audio from WebRTC. The ADB knows about number of
+    // channels; hence we can ask for number of samples per channel here.
+    if (audio_device_buffer_->RequestPlayoutData(
+            playout_samples_per_channel_10ms_) ==
+        static_cast<int32_t>(playout_samples_per_channel_10ms_)) {
+      // Append 10ms to the end of the local buffer taking number of channels
+      // into account.
+      const size_t num_elements_10ms =
+          playout_channels_ * playout_samples_per_channel_10ms_;
+      const size_t written_elements = playout_buffer_.AppendData(
+          num_elements_10ms, [&](rtc::ArrayView<int16_t> buf) {
+            const size_t samples_per_channel_10ms =
+                audio_device_buffer_->GetPlayoutData(buf.data());
+            return playout_channels_ * samples_per_channel_10ms;
+          });
+      RTC_DCHECK_EQ(num_elements_10ms, written_elements);
+    } else {
+      // Provide silence if AudioDeviceBuffer::RequestPlayoutData() fails.
+      // Can e.g. happen when an AudioTransport has not been registered.
+      const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+      std::memset(audio_buffer.data(), 0, num_bytes);
+      return;
+    }
+  }
+
+  // Provide the requested number of bytes to the consumer.
+  const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+  memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
+  // Move remaining samples to start of buffer to prepare for next round.
+  memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(),
+          (playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t));
+  playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size());
+  // Cache playout latency for usage in DeliverRecordedData();
+  playout_delay_ms_ = playout_delay_ms;
+}
+
+void FineAudioBuffer::DeliverRecordedData(
+    rtc::ArrayView<const int16_t> audio_buffer,
+    int record_delay_ms) {
+  RTC_DCHECK(IsReadyForRecord());
+  // Always append new data and grow the buffer when needed.
+  record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
+  // Consume samples from buffer in chunks of 10ms until there is not
+  // enough data left. The number of remaining samples in the cache is given by
+  // the new size of the internal `record_buffer_`.
+  const size_t num_elements_10ms =
+      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_);
+    audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
+    audio_device_buffer_->DeliverRecordedData();
+    memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
+            (record_buffer_.size() - num_elements_10ms) * sizeof(int16_t));
+    record_buffer_.SetSize(record_buffer_.size() - num_elements_10ms);
+  }
+}
+
+}  // namespace webrtc