Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index 078aea66a9fd9b5b0c1c6689b6572a062f8bade0..7f0d825610336f64f4986aea0f5fafc48886697f 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -246,6 +246,12 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps, |
uint8_t fraction_loss, |
int64_t rtt, |
int64_t probing_interval_ms) { |
+ // A send stream may be allocated a bitrate of zero if the allocator decides |
+ // to disable it. For now we ignore this decision and keep sending on min |
+ // bitrate. |
+ if (bitrate_bps == 0) { |
+ bitrate_bps = config_.min_bitrate_bps; |
+ } |
RTC_DCHECK_GE(bitrate_bps, |
static_cast<uint32_t>(config_.min_bitrate_bps)); |
// The bitrate allocator might allocate an higher than max configured bitrate |