commit 2ca0b3689f6072c1b3d12eee29b90b9f1b0784b2
author Per Åhgren <peah@webrtc.org> Mon Aug 26 11:21:16 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Aug 26 09:54:48 2019 +0000
tree dc9174c51f20c3471a474fb15070bced5e5ff757
parent 1fda0277296fd28056bccb124e4994f414b5679b

Correct the handling of sample rates that don't scale well into even 10 ms chunks

This CL corrects the way the audio processing module handles sample rates that
don't allow partitioning the data into evenly sized 10 ms chunks, examples
being 22050 Hz and 11025 Hz.

Bug: webrtc:10882
Change-Id: I35d738f8a0e1debc443fe5d473c0d666a7ba8d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150526
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28953}

modules/audio_processing/audio_buffer.cc

diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 76fabf2..5ac4f94 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -46,11 +46,11 @@
                          size_t buffer_num_channels,
                          size_t output_rate,
                          size_t output_num_channels)
-    : AudioBuffer(rtc::CheckedDivExact(static_cast<int>(input_rate), 100),
+    : AudioBuffer(static_cast<int>(input_rate) / 100,
                   input_num_channels,
-                  rtc::CheckedDivExact(static_cast<int>(buffer_rate), 100),
+                  static_cast<int>(buffer_rate) / 100,
                   buffer_num_channels,
-                  rtc::CheckedDivExact(static_cast<int>(output_rate), 100)) {}
+                  static_cast<int>(output_rate) / 100) {}
 
 AudioBuffer::AudioBuffer(size_t input_num_frames,
                          size_t input_num_channels,