1 files changed, 130 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
new file mode 100644
index 0000000000..29bb0b81d8
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2020 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 "audio/channel_send_frame_transformer_delegate.h"
+
+#include <utility>
+
+namespace webrtc {
+namespace {
+
+class TransformableOutgoingAudioFrame : public TransformableFrameInterface {
+ public:
+  TransformableOutgoingAudioFrame(AudioFrameType frame_type,
+                                  uint8_t payload_type,
+                                  uint32_t rtp_timestamp,
+                                  uint32_t rtp_start_timestamp,
+                                  const uint8_t* payload_data,
+                                  size_t payload_size,
+                                  int64_t absolute_capture_timestamp_ms,
+                                  uint32_t ssrc)
+      : frame_type_(frame_type),
+        payload_type_(payload_type),
+        rtp_timestamp_(rtp_timestamp),
+        rtp_start_timestamp_(rtp_start_timestamp),
+        payload_(payload_data, payload_size),
+        absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
+        ssrc_(ssrc) {}
+  ~TransformableOutgoingAudioFrame() override = default;
+  rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    payload_.SetData(data.data(), data.size());
+  }
+  uint32_t GetTimestamp() const override {
+    return rtp_timestamp_ + rtp_start_timestamp_;
+  }
+  uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+
+  AudioFrameType GetFrameType() const { return frame_type_; }
+  uint8_t GetPayloadType() const override { return payload_type_; }
+  int64_t GetAbsoluteCaptureTimestampMs() const {
+    return absolute_capture_timestamp_ms_;
+  }
+  Direction GetDirection() const override { return Direction::kSender; }
+
+ private:
+  AudioFrameType frame_type_;
+  uint8_t payload_type_;
+  uint32_t rtp_timestamp_;
+  uint32_t rtp_start_timestamp_;
+  rtc::Buffer payload_;
+  int64_t absolute_capture_timestamp_ms_;
+  uint32_t ssrc_;
+};
+}  // namespace
+
+ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
+    SendFrameCallback send_frame_callback,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    rtc::TaskQueue* encoder_queue)
+    : send_frame_callback_(send_frame_callback),
+      frame_transformer_(std::move(frame_transformer)),
+      encoder_queue_(encoder_queue) {}
+
+void ChannelSendFrameTransformerDelegate::Init() {
+  frame_transformer_->RegisterTransformedFrameCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this));
+}
+
+void ChannelSendFrameTransformerDelegate::Reset() {
+  frame_transformer_->UnregisterTransformedFrameCallback();
+  frame_transformer_ = nullptr;
+
+  MutexLock lock(&send_lock_);
+  send_frame_callback_ = SendFrameCallback();
+}
+
+void ChannelSendFrameTransformerDelegate::Transform(
+    AudioFrameType frame_type,
+    uint8_t payload_type,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_start_timestamp,
+    const uint8_t* payload_data,
+    size_t payload_size,
+    int64_t absolute_capture_timestamp_ms,
+    uint32_t ssrc) {
+  frame_transformer_->Transform(
+      std::make_unique<TransformableOutgoingAudioFrame>(
+          frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
+          payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
+}
+
+void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  MutexLock lock(&send_lock_);
+  if (!send_frame_callback_)
+    return;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this);
+  encoder_queue_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        delegate->SendFrame(std::move(frame));
+      });
+}
+
+void ChannelSendFrameTransformerDelegate::SendFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) const {
+  MutexLock lock(&send_lock_);
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  RTC_CHECK_EQ(frame->GetDirection(),
+               TransformableFrameInterface::Direction::kSender);
+  if (!send_frame_callback_)
+    return;
+  auto* transformed_frame =
+      static_cast<TransformableOutgoingAudioFrame*>(frame.get());
+  send_frame_callback_(transformed_frame->GetFrameType(),
+                       transformed_frame->GetPayloadType(),
+                       transformed_frame->GetTimestamp() -
+                           transformed_frame->GetStartTimestamp(),
+                       transformed_frame->GetData(),
+                       transformed_frame->GetAbsoluteCaptureTimestampMs());
+}
+
+}  // namespace webrtc