Add MixingAudioDeviceModule and loopback mixing transport

Introduces infrastructure to mix system-audio loopback into the outgoing
microphone stream during a call, on both Linux and Windows.

New types (in webrtc_create_adm.cpp, anonymous namespace):

  LoopbackCollector
    Thread-safe mono ring buffer (max 2 s at 48 kHz). Loopback capture
    threads call pushSamples(); MixingAudioTransport calls readAndMix()
    to saturating-add loopback audio into the mic frame.

  DirectLoopbackCapture  (Linux only, guarded by WEBRTC_LINUX)
    Background std::thread that opens the PulseAudio monitor source via
    alcCaptureOpenDevice (stereo preferred, mono fallback), feeds decoded
    frames into LoopbackCollector, and stops cleanly on destruction.
    findMonitorDevice() prefers the monitor that matches the current
    default playback sink.

  MixingAudioTransport
    webrtc::AudioTransport decorator. When mixing is enabled it copies the
    mic buffer, calls LoopbackCollector::readAndMix, then forwards the
    blended frame to the inner transport. Disabled path is zero-overhead.

  MixingAudioDeviceModule  (details namespace)
    webrtc::AudioDeviceModule wrapper. Installs MixingAudioTransport in
    RegisterAudioCallback(), starts/stops DirectLoopbackCapture (Linux) or
    a dedicated loopback ADM + LoopbackAdmTransport (Windows) when
    setLoopbackEnabled() is toggled via MixingAudioControl.

New public API (webrtc_create_adm.h):

  MixingAudioControl
    Shared handle that survives ADM recreation. setLoopbackEnabled() /
    loopbackEnabled() let callers toggle mixing at any time; the control
    re-applies pending state whenever a new MixingAudioDeviceModule
    attaches itself.

  MixingAudioDeviceModuleCreator(innerCreator, control)
    Returns a creator lambda that wraps any ADM (e.g. the OpenAL ADM)
    inside a MixingAudioDeviceModule.

Author: paterkleomenis