doc: Add release notes about changes and API breakage

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+Release 0.2
+-----------
+
+Updated AudioProcessing code to be more current.
+
+Contains API breaking changes.
+
+Upstream changes include:
+
+  * Rewritten AGC and voice activity detection
+  * Intelligibility enhancer
+  * Extended AEC filter
+  * Beamformer
+  * Transient suppressor
+  * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
+
+API changes:
+
+  * We no longer include a top-level audio_processing.h. The webrtc tree format
+    is used, so use webrtc/modules/audio_processing/include/audio_processing.h
+  * The top-level module_common_types.h has also been moved to
+    webrtc/modules/interface/module_common_types.h
+  * C++11 support is now required while compiling client code
+  * AudioProcessing::Create() does not take any arguments any more
+  * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
+  * Stream parameters are now configured via StreamConfig and ProcessingConfig
+    rather than set_sample_rate(), set_num_channels(), etc.
+  * AudioFrame field names have changed
+  * Use config API for newer audio processing options
+  * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
+    when using the intelligibility enhancer
+  * GainControl::set_analog_level_limits() is broken. The AGC implementation
+    hard codes 0-255 as the volume range
+
+Other notes:
+
+  * The new audio processing parameters are not all tested, and a few are not
+    enabled upstream (in Chromium) either
+  * The rewritten AGC appears to be less sensitive, and it might make sense to
+    initialise the capture volume to something reasonable (33% or 50%, for
+    example) to make sure there is sufficient energy in the stream to trigger
+    the AGC mechanism
+
+
 Release 0.1
 -----------