Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/test/call_test.cc

Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 833e53c4652288b3a268103432c07214876d08d5..6a2c390261f684ef4d70e1284c0b2d6293e372a9 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -15,6 +15,7 @@
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -37,6 +38,7 @@ CallTest::CallTest()
       num_audio_streams_(0),
       num_flexfec_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      encoder_factory_(CreateBuiltinAudioEncoderFactory()),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
@@ -214,8 +216,9 @@ void CallTest::CreateSendConfig(size_t num_video_streams,
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.voe_channel_id = voe_send_.channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
-    audio_send_config_.send_codec_spec.codec_inst =
-        CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
+    audio_send_config_.send_codec_spec.payload_type = kAudioSendPayloadType;
+    audio_send_config_.send_codec_spec.format = {"ISAC", 16000, 1};
+    audio_send_config_.encoder_factory = encoder_factory_;
   }
 
   // TODO(brandtr): Update this when we support multistream protection.