From 9a9b462e16f1dc4f3c4dbf34688c6beab71981d0 Mon Sep 17 00:00:00 2001
From: Michael Klingbeil <klingm@amazon.com>
Date: Tue, 29 Aug 2023 14:50:49 -0400
Subject: [PATCH] Add Opus FEC options to rtp_encode tool

Bug: None
Change-Id: I7be70951c20069207963b0fa43564c4008eda870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318220
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40668}
---
 AUTHORS                                        | 1 +
 modules/audio_coding/neteq/tools/rtp_encode.cc | 6 +++++-
 2 files changed, 6 insertions(+), 1 deletion(-)

diff --git a/AUTHORS b/AUTHORS
index eec1a87a8c..eb393ef057 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -80,6 +80,7 @@ Maksim Sisov <msisov@igalia.com>
 Maxim Pavlov <pavllovmax@gmail.com>
 Maxim Potapov <vopatop.skam@gmail.com>
 Michael Iedema <michael@kapsulate.com>
+Michael Klingbeil <klingm@amazon.com>
 Michał Zarach <michalzaq12@gmail.com>
 Michel Promonet <michel.promonet.1@gmail.com>
 Miguel Paris <mparisdiaz@gmail.com>
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index e7fca1b5c5..0dfa8764b2 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -42,7 +42,7 @@ ABSL_FLAG(int,
           frame_len,
           0,
           "Frame length in ms; 0 indicates codec default value");
-ABSL_FLAG(int, bitrate, 0, "Bitrate in kbps; 0 indicates codec default value");
+ABSL_FLAG(int, bitrate, 0, "Bitrate in bps; 0 indicates codec default value");
 ABSL_FLAG(int,
           payload_type,
           -1,
@@ -54,6 +54,8 @@ ABSL_FLAG(int,
 ABSL_FLAG(int, ssrc, 0, "SSRC to write to the RTP header");
 ABSL_FLAG(bool, dtx, false, "Use DTX/CNG");
 ABSL_FLAG(int, sample_rate, 48000, "Sample rate of the input file");
+ABSL_FLAG(bool, fec, false, "Use Opus FEC");
+ABSL_FLAG(int, expected_loss, 0, "Expected packet loss percentage");
 
 namespace webrtc {
 namespace test {
@@ -202,6 +204,7 @@ std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
         config.bitrate_bps = absl::GetFlag(FLAGS_bitrate);
       }
       config.dtx_enabled = absl::GetFlag(FLAGS_dtx);
+      config.fec_enabled = absl::GetFlag(FLAGS_fec);
       RTC_CHECK(config.IsOk());
       return AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
     }
@@ -309,6 +312,7 @@ int RunRtpEncode(int argc, char* argv[]) {
   const int timestamp_rate_hz = codec->RtpTimestampRateHz();
   auto acm(AudioCodingModule::Create());
   acm->SetEncoder(std::move(codec));
+  acm->SetPacketLossRate(absl::GetFlag(FLAGS_expected_loss));
 
   // Open files.
   printf("Input file: %s\n", args[1]);