From 8038d42749e9edd52487baea050acda6f604bf91 Mon Sep 17 00:00:00 2001
From: "henrik.lundin@webrtc.org" <henrik.lundin@webrtc.org>
Date: Tue, 11 Nov 2014 08:38:24 +0000
Subject: [PATCH] Follow-up fixes for G722

This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 talk/media/webrtc/webrtcvoiceengine.cc | 17 +++++++++--------
 talk/media/webrtc/webrtcvoiceengine.h  |  2 +-
 2 files changed, 10 insertions(+), 9 deletions(-)

diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 74b316353..c97ed7509 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -74,6 +74,7 @@ static const CodecPref kCodecPrefs[] = {
   { "ISAC",   32000,  1, 104, true },
   { "CELT",   32000,  1, 109, true },
   { "CELT",   32000,  2, 110, true },
+  // G722 should be advertised as 8000 Hz because of the RFC "bug".
   { "G722",   8000,   1, 9,   false },
   { "ILBC",   8000,   1, 102, false },
   { "PCMU",   8000,   1, 0,   false },
@@ -503,7 +504,7 @@ void WebRtcVoiceEngine::ConstructCodecs() {
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
   for (int i = 0; i < ncodecs; ++i) {
     webrtc::CodecInst voe_codec;
-    if (GetVoeCodec(i, voe_codec)) {
+    if (GetVoeCodec(i, &voe_codec)) {
       // Skip uncompressed formats.
       if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
         continue;
@@ -553,13 +554,13 @@ void WebRtcVoiceEngine::ConstructCodecs() {
   std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
 }
 
-bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst& codec) {
-  if (voe_wrapper_->codec()->GetCodec(index, codec) != -1) {
-    // Change the sample rate of G722 to 8000 to match SDP.
-    MaybeFixupG722(&codec, 8000);
-    return true;
+bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
+  if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) {
+    return false;
   }
-  return false;
+  // Change the sample rate of G722 to 8000 to match SDP.
+  MaybeFixupG722(codec, 8000);
+  return true;
 }
 
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
@@ -1246,7 +1247,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
   for (int i = 0; i < ncodecs; ++i) {
     webrtc::CodecInst voe_codec;
-    if (GetVoeCodec(i, voe_codec)) {
+    if (GetVoeCodec(i, &voe_codec)) {
       AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
                        voe_codec.rate, voe_codec.channels, 0);
       bool multi_rate = IsCodecMultiRate(voe_codec);
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 34b9f3c57..67fadc5e3 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -199,7 +199,7 @@ class WebRtcVoiceEngine
 
   void Construct();
   void ConstructCodecs();
-  bool GetVoeCodec(int index, webrtc::CodecInst& codec);
+  bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   bool EnsureSoundclipEngineInit();
   void SetTraceFilter(int filter);