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Change default adaptive ptime min bitrate to 16kbps.

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This is to allow FEC to be encoded at the lowest bitrate.

Bug: chromium:1086942
Change-Id: I1d30276a9a2aaa80016250dc786d5d867ba6cd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212501
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33539}
This commit is contained in:
Jakob Ivarsson 2021-03-23 10:54:00 +01:00 committed by Commit Bot
parent f0adf38d51
commit 7cbe88767b
2 changed files with 4 additions and 2 deletions
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4
media/engine/webrtc_voice_engine.cc
View File

@ -211,7 +211,9 @@ bool IsEnabled(const webrtc::WebRtcKeyValueConfig& config,
struct AdaptivePtimeConfig {
bool enabled = false;
webrtc::DataRate min_payload_bitrate = webrtc::DataRate::KilobitsPerSec(16);
webrtc::DataRate min_encoder_bitrate = webrtc::DataRate::KilobitsPerSec(12);
// Value is chosen to ensure FEC can be encoded, see LBRR_WB_MIN_RATE_BPS in
// libopus.
webrtc::DataRate min_encoder_bitrate = webrtc::DataRate::KilobitsPerSec(16);
bool use_slow_adaptation = true;
absl::optional<std::string> audio_network_adaptor_config;

2
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -1218,7 +1218,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) {
parameters.encodings[0].adaptive_ptime = true;
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX));
EXPECT_EQ(12000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
parameters.encodings[0].adaptive_ptime = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());