Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
 #include "webrtc/test/call_test.h"
 
 #include <algorithm>
 
 #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"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
       event_log_(RtcEventLog::CreateNull()),
       video_send_config_(nullptr),
       video_send_stream_(nullptr),
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(1),
       num_audio_streams_(0),
       num_flexfec_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      encoder_factory_(CreateBuiltinAudioEncoderFactory()),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
 CallTest::~CallTest() {
 }
 
 void CallTest::RunBaseTest(BaseTest* test) {
   num_video_streams_ = test->GetNumVideoStreams();
   num_audio_streams_ = test->GetNumAudioStreams();
   num_flexfec_streams_ = test->GetNumFlexfecStreams();
@@ skipping 164 matching lines @@
     for (size_t i = 0; i < num_video_streams; ++i)
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
   }
 
   if (num_audio_streams > 0) {
     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, "OPUS", 48000, 960, 2, 64000};
+    audio_send_config_.send_codec_spec =
+        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+            {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}});
+    audio_send_config_.encoder_factory = encoder_factory_;
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   if (num_flexfec_streams > 0) {
     video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
     video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
 
@@ skipping 302 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc