This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.
Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.
Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
R=turaj@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21519007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
