Pass SdpAudioFormat through Channel, without converting to CodecInst

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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.
  */
 
 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 
 #include <list>
 #include <map>
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace cricket {
 
 static const int kOpusBandwidthNb = 4000;
 static const int kOpusBandwidthMb = 6000;
 static const int kOpusBandwidthWb = 8000;
 static const int kOpusBandwidthSwb = 12000;
 static const int kOpusBandwidthFb = 20000;
 
@@ skipping 101 matching lines @@
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
   WEBRTC_STUB(SetSendCodec, (int channel, const webrtc::CodecInst& codec));
   WEBRTC_STUB(GetSendCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
   WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetRecPayloadType, (int channel,
                                   const webrtc::CodecInst& codec)) {
+    return SetRecPayloadType(channel, codec.pltype, CodecInstToSdp(codec));
+  }
+  WEBRTC_FUNC(SetRecPayloadType,
+              (int channel,
+               int payload_type,
+               const webrtc::SdpAudioFormat& format)) {
     WEBRTC_CHECK_CHANNEL(channel);
     Channel* ch = channels_[channel];
     // Check if something else already has this slot.
-    if (codec.pltype != -1) {
+    if (payload_type != -1) {
       for (std::vector<webrtc::CodecInst>::iterator it =
           ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
-        if (it->pltype == codec.pltype &&
-            _stricmp(it->plname, codec.plname) != 0) {
+        if (it->pltype == payload_type &&
+            _stricmp(it->plname, format.name.c_str()) != 0) {
           return -1;
         }
       }
     }
     // Otherwise try to find this codec and update its payload type.
     int result = -1;  // not found
-    for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
-         it != ch->recv_codecs.end(); ++it) {
-      if (strcmp(it->plname, codec.plname) == 0 &&
-          it->plfreq == codec.plfreq &&
-          it->channels == codec.channels) {
-        it->pltype = codec.pltype;
+    for (auto& codec : ch->recv_codecs) {
+      if (CodecInstToSdp(codec) == format) {
+        codec.pltype = payload_type;
         result = 0;
       }
     }
     return result;
   }
   WEBRTC_STUB(SetSendCNPayloadType, (int channel, int type,
                                      webrtc::PayloadFrequencies frequency));
   WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     Channel* ch = channels_[channel];
@@ skipping 176 matching lines @@
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   webrtc::AudioProcessing* apm_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_