Adding an extra C-tor to AudioBuffer

This CL adds an extra C-tor to AudioBuffer, which omits the unused 6th parameter of the preexisting C-tor. Once the new C-tor is adopted in upstream code, a CL will be created that removes the old C-tor.

Bug: webrtc:42221468
Change-Id: I1e38ce3202e333f49c4a5f726a27c2919be6b510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/428901
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Auto-Submit: Per Åhgren <peah@webrtc.org>
Owners-Override: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46398}
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 7f25dba..bf559cf 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -47,6 +47,17 @@
                          size_t buffer_num_channels,
                          size_t output_rate,
                          size_t /* output_num_channels */)
+    : AudioBuffer(input_rate,
+                  input_num_channels,
+                  buffer_rate,
+                  buffer_num_channels,
+                  output_rate) {}
+
+AudioBuffer::AudioBuffer(size_t input_rate,
+                         size_t input_num_channels,
+                         size_t buffer_rate,
+                         size_t buffer_num_channels,
+                         size_t output_rate)
     : input_num_frames_(static_cast<int>(input_rate) / 100),
       input_num_channels_(input_num_channels),
       buffer_num_frames_(static_cast<int>(buffer_rate) / 100),
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 6f11a70..5a1c812 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -37,6 +37,7 @@
   static const int kSplitBandSize = 160;
   // TODO(tommi): Remove this (`AudioBuffer::kMaxSampleRate`) constant.
   static const int kMaxSampleRate = kMaxSampleRateHz;
+
   AudioBuffer(size_t input_rate,
               size_t input_num_channels,
               size_t buffer_rate,
@@ -44,6 +45,12 @@
               size_t output_rate,
               size_t output_num_channels);
 
+  AudioBuffer(size_t input_rate,
+              size_t input_num_channels,
+              size_t buffer_rate,
+              size_t buffer_num_channels,
+              size_t output_rate);
+
   virtual ~AudioBuffer();
 
   AudioBuffer(const AudioBuffer&) = delete;