zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

AnalyzeReverseStream with StreamConfig

Browse Source 
Adding a version of AnalyzeReverseStream with audio parameters
described by StreamConfig. This is part of preparations for
multichannel APM in Chromium.

Bug: webrtc:10913
Change-Id: I7c4650eab8bd7fcdec970a7e4a8fa203f09bed9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157897
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29573}
This commit is contained in:
Gustaf Ullberg
2019-10-22 15:21:31 +02:00
committed by Commit Bot
parent e76b3abf61
commit 8c51f2e9cd
4 changed files with 19 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
modules/audio_processing/include/audio_processing.h
View File

@ -593,6 +593,12 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
int sample_rate_hz,
ChannelLayout layout) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
// of |data| points to a channel buffer, arranged according to
// |reverse_config|.
virtual int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
// |data| points to a channel buffer, arranged according to |reverse_config|.
virtual int ProcessReverseStream(const float* const* src,