#include <string>
#include <vector>
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/voice_engine/transmit_mixer.h"
|
| | webrtc::test::TEST (AudioSendStreamTest, ConfigToString) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, ConstructDestruct) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, SendTelephoneEvent) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, SetMuted) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, GetStats) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, GetStatsTypingNoiseDetected) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, SendCodecAppliesConfigParams) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, SendCodecCanApplyVad) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) |
| |
| | webrtc::test::TEST (AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) |
| |
