From 2ca0b3689f6072c1b3d12eee29b90b9f1b0784b2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Per=20=C3=85hgren?= Date: Mon, 26 Aug 2019 11:21:16 +0200 Subject: [PATCH] Correct the handling of sample rates that don't scale well into even 10 ms chunks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 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 Commit-Queue: Per Åhgren Cr-Commit-Position: refs/heads/master@{#28953} --- modules/audio_processing/audio_buffer.cc | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc index 76fabf2664..5ac4f94d30 100644 --- a/modules/audio_processing/audio_buffer.cc +++ b/modules/audio_processing/audio_buffer.cc @@ -46,11 +46,11 @@ AudioBuffer::AudioBuffer(size_t input_rate, size_t buffer_num_channels, size_t output_rate, size_t output_num_channels) - : AudioBuffer(rtc::CheckedDivExact(static_cast(input_rate), 100), + : AudioBuffer(static_cast(input_rate) / 100, input_num_channels, - rtc::CheckedDivExact(static_cast(buffer_rate), 100), + static_cast(buffer_rate) / 100, buffer_num_channels, - rtc::CheckedDivExact(static_cast(output_rate), 100)) {} + static_cast(output_rate) / 100) {} AudioBuffer::AudioBuffer(size_t input_num_frames, size_t input_num_channels,