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 496a871d85de84735c89a3cff3e88eb6f1223fd9..55e234f940b8aa0538eb39e88d2f0ea78ad68c86 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -106,6 +106,8 @@ struct ConfigHelper { |
// Use ISAC as default codec so as to prevent unnecessary |voice_engine_| |
// calls from the default ctor behavior. |
stream_config_.send_codec_spec.codec_inst = kIsacCodec; |
+ stream_config_.min_bitrate_bps = 10000; |
+ stream_config_.max_bitrate_bps = 65000; |
} |
AudioSendStream::Config& config() { return stream_config_; } |
@@ -403,5 +405,27 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) { |
helper.bitrate_allocator(), helper.event_log()); |
} |
+TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
+ ConfigHelper helper; |
+ internal::AudioSendStream send_stream( |
+ helper.config(), helper.audio_state(), helper.worker_queue(), |
+ helper.packet_router(), helper.congestion_controller(), |
+ helper.bitrate_allocator(), helper.event_log()); |
+ EXPECT_CALL(*helper.channel_proxy(), |
+ SetBitrate(helper.config().max_bitrate_bps, _)); |
+ send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, |
+ 6000); |
+} |
+ |
+TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { |
+ ConfigHelper helper; |
+ internal::AudioSendStream send_stream( |
+ helper.config(), helper.audio_state(), helper.worker_queue(), |
+ helper.packet_router(), helper.congestion_controller(), |
+ helper.bitrate_allocator(), helper.event_log()); |
+ EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); |
+ send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); |
+} |
+ |
} // namespace test |
} // namespace webrtc |