Index: webrtc/audio/test/low_bandwidth_audio_test.cc |
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc |
index 98cfa703c3953e1162edd898689a72fd61b6a001..f9e6e2b9d1fdaab12f0bfdc7ccfec7f8b486df26 100644 |
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc |
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc |
@@ -97,8 +97,12 @@ test::PacketTransport* AudioQualityTest::CreateReceiveTransport() { |
void AudioQualityTest::ModifyAudioConfigs( |
AudioSendStream::Config* send_config, |
std::vector<AudioReceiveStream::Config>* receive_configs) { |
- send_config->send_codec_spec.codec_inst = webrtc::CodecInst{ |
- test::CallTest::kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000}; |
+ // Large bitrate by default. |
+ const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2, |
+ {{"stereo", "1"}}); |
+ send_config->send_codec_spec = |
+ rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
+ {test::CallTest::kAudioSendPayloadType, kDefaultFormat}); |
} |
void AudioQualityTest::PerformTest() { |
@@ -130,14 +134,15 @@ TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) { |
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{ |
- test::CallTest::kAudioSendPayloadType, // pltype |
- "OPUS", // plname |
- 48000, // plfreq |
- 2880, // pacsize |
- 1, // channels |
- 6000 // rate bits/sec |
- }; |
+ send_config->send_codec_spec = |
+ rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
+ {test::CallTest::kAudioSendPayloadType, |
+ {"OPUS", |
+ 48000, |
+ 2, |
+ {{"maxaveragebitrate", "6000"}, |
+ {"ptime", "60"}, |
+ {"stereo", "1"}}}}); |
} |
FakeNetworkPipe::Config GetNetworkPipeConfig() override { |