Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 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 <algorithm>
 #include <limits>
 #include <memory>
 #include <string>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call/call.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
@@ skipping 182 matching lines @@
   receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
   CreateSendConfig(1, 0, 0, &video_send_transport);
   CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
-  audio_send_config.send_codec_spec.codec_inst =
-      CodecInst{103, "ISAC", 16000, 480, 1, 32000};
+  audio_send_config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {103, {"ISAC", 16000, 1}});
+  audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
   AudioSendStream* audio_send_stream =
       sender_call_->CreateAudioSendStream(audio_send_config);
 
   video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   if (fec == FecMode::kOn) {
     video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
     video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
     video_receive_configs_[0].rtp.ulpfec.red_payload_type = kRedPayloadType;
     video_receive_configs_[0].rtp.ulpfec.ulpfec_payload_type =
         kUlpfecPayloadType;
@@ skipping 527 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc