From 2e03c6611927af6ed08cf967b95342744ff329b6 Mon Sep 17 00:00:00 2001
From: minyue <minyue@webrtc.org>
Date: Wed, 1 Feb 2017 17:31:11 -0800
Subject: [PATCH] Adding build switch for Opus that supports 120ms ptime.

BUG=webrtc:7097

TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o

Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
---
 webrtc/media/BUILD.gn                                    | 6 ++++++
 webrtc/media/engine/webrtcvoiceengine.cc                 | 5 +++++
 webrtc/modules/audio_coding/BUILD.gn                     | 2 ++
 webrtc/modules/audio_coding/acm2/acm_codec_database.cc   | 4 ++++
 webrtc/modules/audio_coding/audio_coding.gni             | 5 +++++
 webrtc/modules/audio_coding/codecs/opus/opus_interface.c | 5 +++++
 webrtc/webrtc.gni                                        | 4 ++++
 7 files changed, 31 insertions(+)

diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 06aed45662..c223068c20 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -161,6 +161,12 @@ rtc_static_library("rtc_media") {
     defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
   }
 
+  if (rtc_opus_support_120ms_ptime) {
+    defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+  } else {
+    defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+  }
+
   include_dirs = []
   if (rtc_build_libyuv) {
     deps += [ "$rtc_libyuv_dir" ]
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index b9a725b0a0..1ed1e8b993 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -498,7 +498,12 @@ class WebRtcVoiceCodecs final {
 };
 
 const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60, 120},
+     kOpusMaxBitrateBps},
+#else
     {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
+#endif
     {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
     {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
     // G722 should be advertised as 8000 Hz because of the RFC "bug".
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 1d3c2d1016..59c166db99 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -872,6 +872,8 @@ rtc_source_set("webrtc_opus_c") {
     "codecs/opus/opus_interface.h",
   ]
 
+  defines = audio_coding_defines
+
   if (rtc_build_opus) {
     public_deps = [
       rtc_opus_dir,
diff --git a/webrtc/modules/audio_coding/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
index 4f5b263c24..0fae75b91b 100644
--- a/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
@@ -151,7 +151,11 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
     // Opus supports frames shorter than 10ms,
     // but it doesn't help us to use them.
     // Mono and stereo.
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+    {5, {480, 960, 1920, 2880, 5760}, 0, 2},
+#else
     {4, {480, 960, 1920, 2880}, 0, 2},
+#endif
 #endif
     // Comfort noise for three different sampling frequencies.
     {1, {240}, 240, 1},
diff --git a/webrtc/modules/audio_coding/audio_coding.gni b/webrtc/modules/audio_coding/audio_coding.gni
index 0f3a75fbac..41dcf0044c 100644
--- a/webrtc/modules/audio_coding/audio_coding.gni
+++ b/webrtc/modules/audio_coding/audio_coding.gni
@@ -15,6 +15,11 @@ if (rtc_include_ilbc) {
 if (rtc_include_opus) {
   audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
 }
+if (rtc_opus_support_120ms_ptime) {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+} else {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+}
 if (!build_with_mozilla) {
   if (current_cpu == "arm") {
     audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index e2b4fdbeee..d544d850a9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -17,8 +17,13 @@
 #include <string.h>
 
 enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
   /* Maximum supported frame size in WebRTC is 60 ms. */
   kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
 
   /* The format allows up to 120 ms frames. Since we don't control the other
    * side, we must allow for packets of that size. NetEq is currently limited
diff --git a/webrtc/webrtc.gni b/webrtc/webrtc.gni
index 53dcae2b7c..4c34eae192 100644
--- a/webrtc/webrtc.gni
+++ b/webrtc/webrtc.gni
@@ -17,6 +17,10 @@ declare_args() {
   # Disable this to avoid building the Opus audio codec.
   rtc_include_opus = true
 
+  # Enable this if the Opus version upon which WebRTC is built supports direct
+  # encoding of 120 ms packets.
+  rtc_opus_support_120ms_ptime = false
+
   # Enable this to let the Opus audio codec change complexity on the fly.
   rtc_opus_variable_complexity = false