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Unified Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 2247213005: Fixing config for Audio BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebasing Created 4 years, 2 months ago
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Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index a2832ded00cb330d86ffae8805955326cb1033a2..140f3cd58495d55cdd0dac89b2c63e11d8390b9f 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -219,8 +219,8 @@ TEST(AudioSendStreamTest, ConfigToString) {
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
config.rtp.c_name = kCName;
config.voe_channel_id = kChannelId;
- config.min_bitrate_kbps = 12;
- config.max_bitrate_kbps = 34;
+ config.min_bitrate_bps = 12000;
+ config.max_bitrate_bps = 34000;
config.send_codec_spec.nack_enabled = true;
config.send_codec_spec.transport_cc_enabled = false;
config.send_codec_spec.enable_codec_fec = true;
@@ -233,7 +233,7 @@ TEST(AudioSendStreamTest, ConfigToString) {
"{rtp: {ssrc: 1234, extensions: [{uri: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
"nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
- "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
+ "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
"32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: "
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