From b9d468573a051bceb3d8fb2fbbb6c7d7cab77bbe Mon Sep 17 00:00:00 2001
From: Philipp Hancke <philipp.hancke@googlemail.com>
Date: Tue, 14 Apr 2020 15:26:05 +0200
Subject: [PATCH] insertable streams: include rtp_timestamp offset for audio
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

includes the (random) rtp start offset in the timestamp passed to the frame transformer callback

Bug: chromium:1069278
Change-Id: I7d10130404d93df7cee3b8f87a0b780801a51415
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173329
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31067}
---
 audio/channel_send.cc                         |  5 ++--
 ...channel_send_frame_transformer_delegate.cc | 23 +++++++++++++------
 .../channel_send_frame_transformer_delegate.h |  1 +
 3 files changed, 20 insertions(+), 9 deletions(-)

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 21f36aaa17..d2d4d2be6d 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -389,8 +389,9 @@ int32_t ChannelSend::SendData(AudioFrameType frameType,
     // Asynchronously transform the payload before sending it. After the payload
     // is transformed, the delegate will call SendRtpAudio to send it.
     frame_transformer_delegate_->Transform(
-        frameType, payloadType, rtp_timestamp, payloadData, payloadSize,
-        absolute_capture_timestamp_ms, _rtpRtcpModule->SSRC());
+        frameType, payloadType, rtp_timestamp, _rtpRtcpModule->StartTimestamp(),
+        payloadData, payloadSize, absolute_capture_timestamp_ms,
+        _rtpRtcpModule->SSRC());
     return 0;
   }
   return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 53df6b0ff0..53b573eb8b 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -20,6 +20,7 @@ class TransformableAudioFrame : public TransformableFrameInterface {
   TransformableAudioFrame(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,
@@ -27,6 +28,7 @@ class TransformableAudioFrame : public TransformableFrameInterface {
       : 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) {}
@@ -35,7 +37,10 @@ class TransformableAudioFrame : public TransformableFrameInterface {
   void SetData(rtc::ArrayView<const uint8_t> data) override {
     payload_.SetData(data.data(), data.size());
   }
-  uint32_t GetTimestamp() const override { return rtp_timestamp_; }
+  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_; }
@@ -48,6 +53,7 @@ class TransformableAudioFrame : public TransformableFrameInterface {
   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_;
@@ -79,13 +85,14 @@ 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<TransformableAudioFrame>(
-      frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
-      absolute_capture_timestamp_ms, ssrc));
+      frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
+      payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
 }
 
 void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
@@ -107,10 +114,12 @@ void ChannelSendFrameTransformerDelegate::SendFrame(
   if (!send_frame_callback_)
     return;
   auto* transformed_frame = static_cast<TransformableAudioFrame*>(frame.get());
-  send_frame_callback_(
-      transformed_frame->GetFrameType(), transformed_frame->GetPayloadType(),
-      transformed_frame->GetTimestamp(), transformed_frame->GetData(),
-      transformed_frame->GetAbsoluteCaptureTimestampMs());
+  send_frame_callback_(transformed_frame->GetFrameType(),
+                       transformed_frame->GetPayloadType(),
+                       transformed_frame->GetTimestamp() -
+                           transformed_frame->GetStartTimestamp(),
+                       transformed_frame->GetData(),
+                       transformed_frame->GetAbsoluteCaptureTimestampMs());
 }
 
 }  // namespace webrtc
diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h
index f2655e7c51..5added7b31 100644
--- a/audio/channel_send_frame_transformer_delegate.h
+++ b/audio/channel_send_frame_transformer_delegate.h
@@ -54,6 +54,7 @@ class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
   void 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,