Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index 840ae68d062da6dbca584258a11554c7e717c4c4..b031762b4cecd173e21766e4ab3f91c070a37fd9 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -229,8 +229,7 @@ |
uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps, |
uint8_t fraction_loss, |
- int64_t rtt, |
- int64_t probing_interval_ms) { |
+ int64_t rtt) { |
RTC_DCHECK_GE(bitrate_bps, |
static_cast<uint32_t>(config_.min_bitrate_bps)); |
// The bitrate allocator might allocate an higher than max configured bitrate |
@@ -239,7 +238,7 @@ |
if (bitrate_bps > max_bitrate_bps) |
bitrate_bps = max_bitrate_bps; |
- channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms); |
+ channel_proxy_->SetBitrate(bitrate_bps); |
// The amount of audio protection is not exposed by the encoder, hence |
// always returning 0. |