| 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 {
|
|
|