Add format members to AudioConverter for DCHECKing.

And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_audio/audio_converter.cc

diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index 9e18033..f085ff1 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -43,10 +43,13 @@
 }  // namespace
 
 AudioConverter::AudioConverter(int src_channels, int src_frames,
-                               int dst_channels, int dst_frames) {
+                               int dst_channels, int dst_frames)
+    : src_channels_(src_channels),
+      src_frames_(src_frames),
+      dst_channels_(dst_channels),
+      dst_frames_(dst_frames) {
   CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
-  const int resample_channels = src_channels < dst_channels ? src_channels :
-                                                              dst_channels;
+  const int resample_channels = std::min(src_channels, dst_channels);
 
   // Prepare buffers as needed for intermediate stages.
   if (dst_channels < src_channels)
@@ -66,8 +69,11 @@
                              int dst_channels,
                              int dst_frames,
                              float* const* dst) {
-  DCHECK(dst_channels == src_channels || dst_channels == 1 ||
-         src_channels == 1);
+  DCHECK_EQ(src_channels_, src_channels);
+  DCHECK_EQ(src_frames_, src_frames);
+  DCHECK_EQ(dst_channels_, dst_channels);
+  DCHECK_EQ(dst_frames_, dst_frames);;
+
   if (src_channels == dst_channels && src_frames == dst_frames) {
     // Shortcut copy.
     if (src != dst) {

webrtc/common_audio/audio_converter.h

diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
index df31755e..6365f58 100644
--- a/webrtc/common_audio/audio_converter.h
+++ b/webrtc/common_audio/audio_converter.h
@@ -40,6 +40,10 @@
                float* const* dest);
 
  private:
+  const int src_channels_;
+  const int src_frames_;
+  const int dst_channels_;
+  const int dst_frames_;
   scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
   ScopedVector<PushSincResampler> resamplers_;