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 a296513d9dfc4b1add42931a85c09e100e0028ef..dbc49662e6be7c4ebcf25d2d478fcee9091ec00f 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -105,8 +105,6 @@ |
// 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_; } |
@@ -402,27 +400,5 @@ |
helper.event_log()); |
} |
-TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
- ConfigHelper helper; |
- internal::AudioSendStream send_stream( |
- helper.config(), helper.audio_state(), helper.worker_queue(), |
- 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.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 |