Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 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 "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace {
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
-// Large bitrate by default.
-const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000};
-
 // The best that can be done with PESQ.
 constexpr int kAudioFileBitRate = 16000;
 }
 
 namespace webrtc {
 namespace test {
 
 AudioQualityTest::AudioQualityTest()
     : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
 
@@ skipping 49 matching lines @@
 
 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
   return new test::PacketTransport(nullptr, this,
       test::PacketTransport::kReceiver, MediaType::AUDIO,
       GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
-  send_config->send_codec_spec.codec_inst = kDefaultCodec;
+  // Large bitrate by default.
+  const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
+                                              {{"stereo", "1"}});
+  send_config->send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {120, kDefaultFormat});
 }
 
 void AudioQualityTest::PerformTest() {
   // Wait until the input audio file is done...
   send_audio_device_->WaitForRecordingEnd();
   // and some extra time to account for network delay.
   SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
 }
 
 void AudioQualityTest::OnTestFinished() {
@@ skipping 11 matching lines @@
 
 TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
   AudioQualityTest test;
   RunBaseTest(&test);
 }
 
 
 class Mobile2GNetworkTest : public AudioQualityTest {
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
-    send_config->send_codec_spec.codec_inst = CodecInst{
-        120,     // pltype
-        "OPUS",  // plname
-        48000,   // plfreq
-        2880,    // pacsize
-        1,       // channels
-        6000     // rate bits/sec
-    };
+    send_config->send_codec_spec =
+        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+            {120,
+             {"OPUS",
+              48000,
+              2,
+              {{"maxaveragebitrate", "6000"},
+               {"ptime", "60"},
+               {"stereo", "1"}}}});
   }
 
   FakeNetworkPipe::Config GetNetworkPipeConfig() override {
     FakeNetworkPipe::Config pipe_config;
     pipe_config.link_capacity_kbps = 12;
     pipe_config.queue_length_packets = 1500;
     pipe_config.queue_delay_ms = 400;
     return pipe_config;
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc