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Side by Side Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 2532993002: Revert of Pass time constant to bwe smoothing filter. (Closed)
Patch Set: Created 4 years ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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98 stream_config_.rtp.ssrc = kSsrc; 98 stream_config_.rtp.ssrc = kSsrc;
99 stream_config_.rtp.nack.rtp_history_ms = 200; 99 stream_config_.rtp.nack.rtp_history_ms = 200;
100 stream_config_.rtp.c_name = kCName; 100 stream_config_.rtp.c_name = kCName;
101 stream_config_.rtp.extensions.push_back( 101 stream_config_.rtp.extensions.push_back(
102 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); 102 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
103 stream_config_.rtp.extensions.push_back(RtpExtension( 103 stream_config_.rtp.extensions.push_back(RtpExtension(
104 RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); 104 RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
105 // Use ISAC as default codec so as to prevent unnecessary |voice_engine_| 105 // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
106 // calls from the default ctor behavior. 106 // calls from the default ctor behavior.
107 stream_config_.send_codec_spec.codec_inst = kIsacCodec; 107 stream_config_.send_codec_spec.codec_inst = kIsacCodec;
108 stream_config_.min_bitrate_bps = 10000;
109 stream_config_.max_bitrate_bps = 65000;
110 } 108 }
111 109
112 AudioSendStream::Config& config() { return stream_config_; } 110 AudioSendStream::Config& config() { return stream_config_; }
113 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } 111 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
114 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } 112 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
115 CongestionController* congestion_controller() { 113 CongestionController* congestion_controller() {
116 return &congestion_controller_; 114 return &congestion_controller_;
117 } 115 }
118 BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } 116 BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
119 rtc::TaskQueue* worker_queue() { return &worker_queue_; } 117 rtc::TaskQueue* worker_queue() { return &worker_queue_; }
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395 stream_config.send_codec_spec.cng_plfreq = 8000; 393 stream_config.send_codec_spec.cng_plfreq = 8000;
396 stream_config.send_codec_spec.cng_payload_type = 105; 394 stream_config.send_codec_spec.cng_payload_type = 105;
397 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _)) 395 EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
398 .WillOnce(Return(0)); 396 .WillOnce(Return(0));
399 internal::AudioSendStream send_stream( 397 internal::AudioSendStream send_stream(
400 stream_config, helper.audio_state(), helper.worker_queue(), 398 stream_config, helper.audio_state(), helper.worker_queue(),
401 helper.congestion_controller(), helper.bitrate_allocator(), 399 helper.congestion_controller(), helper.bitrate_allocator(),
402 helper.event_log()); 400 helper.event_log());
403 } 401 }
404 402
405 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
406 ConfigHelper helper;
407 internal::AudioSendStream send_stream(
408 helper.config(), helper.audio_state(), helper.worker_queue(),
409 helper.congestion_controller(), helper.bitrate_allocator(),
410 helper.event_log());
411 EXPECT_CALL(*helper.channel_proxy(),
412 SetBitrate(helper.config().max_bitrate_bps, _));
413 send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
414 6000);
415 }
416
417 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
418 ConfigHelper helper;
419 internal::AudioSendStream send_stream(
420 helper.config(), helper.audio_state(), helper.worker_queue(),
421 helper.congestion_controller(), helper.bitrate_allocator(),
422 helper.event_log());
423 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
424 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
425 }
426
427 } // namespace test 403 } // namespace test
428 } // namespace webrtc 404 } // namespace webrtc
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