henrika 9868042b05 Removes unused APIs from the ADM (part II). ...

Removes:

int32_t SpeakerVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneBoostIsAvailable(bool* available)
int32_t SetMicrophoneBoost(bool enable)
int32_t MicrophoneBoost(bool* enabled)
int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0)
int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS)
int32_t CPULoad(uint16_t* load)
int32_t StartRawOutputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawOutputFileRecording()
int32_t StartRawInputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawInputFileRecording()
int32_t ResetAudioDevice()

BUG=webrtc:7306

Review-Url: https://codereview.webrtc.org/3006803002
Cr-Commit-Position: refs/heads/master@{#19632}

2017-08-31 13:47:32 +00:00

..

audio_coding

Decouple FEC controller's decision from the decision making for frame length.

2017-08-31 12:08:56 +00:00

audio_conference_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_device

Removes unused APIs from the ADM (part II).

2017-08-31 13:47:32 +00:00

audio_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_processing

Run the ClangTidy analyser on the AudioProcessing submodule of WebRTC.

2017-08-31 12:53:44 +00:00

bitrate_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

congestion_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

desktop_capture

Flip IsWindowOnScreen behavior when native APIs fail

2017-08-31 02:43:14 +00:00

include

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

media_file

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

pacing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

remote_bitrate_estimator

Replace remaining gflags usages with rtc_base/flags

2017-08-31 10:21:39 +00:00

rtp_rtcp

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

utility

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_capture

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_coding

Replace remaining gflags usages with rtc_base/flags

2017-08-31 10:21:39 +00:00

video_processing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00