Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/utility.h

Index: webrtc/voice_engine/utility.h
diff --git a/webrtc/voice_engine/utility.h b/webrtc/voice_engine/utility.h
index 87003c42581cd99337e3176fb3b1c2c772d438bc..d84ce2bf6256ec8a629d781fbfc8b05ab7805a1b 100644
--- a/webrtc/voice_engine/utility.h
+++ b/webrtc/voice_engine/utility.h
@@ -26,30 +26,21 @@ namespace voe {
 
 // Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
 // Expects |dst_frame| to have its sample rate and channels members set to the
-// desired values. Updates the samples per channel member accordingly. No other
-// members will be changed.
+// desired values. Updates the |samples_per_channel_|, |timestamp_|,
+// |elapsed_time_ms_| and |ntp_time_ms_| members accordingly.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame);
 
-// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
-// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
-// temporary space and must be of sufficient size to hold the downmixed source
-// audio (recommend using a size of kMaxMonoDataSizeSamples).
-//
-// |dst_af| will have its data and format members (sample rate, channels and
-// samples per channel) set appropriately. No other members will be changed.
-// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
-// it shouldn't be needed.
-void DownConvertToCodecFormat(const int16_t* src_data,
-                              size_t samples_per_channel,
-                              int num_channels,
-                              int sample_rate_hz,
-                              int codec_num_channels,
-                              int codec_rate_hz,
-                              int16_t* mono_buffer,
-                              PushResampler<int16_t>* resampler,
-                              AudioFrame* dst_af);
+// Upmix or downmix and resample the audio in |src_data| to |dst_frame|. Expects
+// dst_frame| to have its sample rate and channels members set to the desired

[Andrew MacDonald, 2015/09/23 17:31:47]

Rather than repeating the comments please write the shared block once above both
and then say "This version..." above each to describe the unique bit.

[aluebs-webrtc, 2015/09/23 19:48:54]

Done.

+// values. Updates the |samples_per_channel_| members accordingly.
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      int num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
 
 void MixWithSat(int16_t target[],
                 int target_channel,