Consolidate audio conversion from Channel and TransmitMixer.

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/webrtc@5843 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 1 insertions, 1 deletions

diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 43cfe99a..b490f489 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -545,7 +545,7 @@ private:
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
-    scoped_array<int16_t> mono_recording_audio_;
+    scoped_ptr<int16_t[]> mono_recording_audio_;
     // Resampler is used when input data is stereo while codec is mono.
     PushResampler input_resampler_;
     uint8_t _audioLevel_dBov;