Fixing config for Audio BWE.
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.
BUG=webrtc:6670
Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 99369a2..6f24ef8 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -83,16 +83,16 @@
// 64-128 kb/s for FB stereo music.
// The current implementation applies the following values to mono signals,
// and multiplies them by 2 for stereo.
-const int kOpusBitrateNb = 12000;
-const int kOpusBitrateWb = 20000;
-const int kOpusBitrateFb = 32000;
+const int kOpusBitrateNbBps = 12000;
+const int kOpusBitrateWbBps = 20000;
+const int kOpusBitrateFbBps = 32000;
// Opus bitrate should be in the range between 6000 and 510000.
-const int kOpusMinBitrate = 6000;
-const int kOpusMaxBitrate = 510000;
+const int kOpusMinBitrateBps = 6000;
+const int kOpusMaxBitrateBps = 510000;
// iSAC bitrate should be <= 56000.
-const int kIsacMaxBitrate = 56000;
+const int kIsacMaxBitrateBps = 56000;
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
@@ -222,18 +222,19 @@
}
if (bitrate <= 0) {
if (max_playback_rate <= 8000) {
- bitrate = kOpusBitrateNb;
+ bitrate = kOpusBitrateNbBps;
} else if (max_playback_rate <= 16000) {
- bitrate = kOpusBitrateWb;
+ bitrate = kOpusBitrateWbBps;
} else {
- bitrate = kOpusBitrateFb;
+ bitrate = kOpusBitrateFbBps;
}
if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
bitrate *= 2;
}
- } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
- bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
+ } else if (bitrate < kOpusMinBitrateBps || bitrate > kOpusMaxBitrateBps) {
+ bitrate = (bitrate < kOpusMinBitrateBps) ? kOpusMinBitrateBps
+ : kOpusMaxBitrateBps;
std::string rate_source =
use_param ? "Codec parameter \"maxaveragebitrate\"" :
"Supplied Opus bitrate";
@@ -478,9 +479,9 @@
};
const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[11] = {
- {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrate},
- {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrate},
- {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrate},
+ {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
+ {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
+ {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
// G722 should be advertised as 8000 Hz because of the RFC "bug".
{kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
{kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
@@ -489,8 +490,7 @@
{kCnCodecName, 32000, 1, 106, false, {}},
{kCnCodecName, 16000, 1, 105, false, {}},
{kCnCodecName, 8000, 1, 13, false, {}},
- {kDtmfCodecName, 8000, 1, 126, false, {}}
-};
+ {kDtmfCodecName, 8000, 1, 126, false, {}}};
rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
int rtp_max_bitrate_bps,
@@ -1392,8 +1392,8 @@
"Enabled") {
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
- config_.min_bitrate_kbps = kOpusMinBitrate;
- config_.max_bitrate_kbps = kOpusBitrateFb;
+ config_.min_bitrate_bps = kOpusMinBitrateBps;
+ config_.max_bitrate_bps = kOpusBitrateFbBps;
}
stream_ = call_->CreateAudioSendStream(config_);
RTC_CHECK(stream_);
