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ACM: Corrected temporary buffer size

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This CL corrects the temporary buffers size in the
pre-processing of the capture audio before encoding.

As part of this it removes the ACM-specific hardcoding
of the size and instead ensures that the size of the
temporary buffer matches that of the AudioFrame.

Bug: webrtc:11242
Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30775}
This commit is contained in:
Per Åhgren
2020-03-12 11:53:30 +01:00
committed by Commit Bot
parent c71be24c82
commit d82a02c837
3 changed files with 20 additions and 9 deletions
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2
modules/audio_coding/include/audio_coding_module.h
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@ -33,8 +33,6 @@ class AudioEncoder;
class AudioFrame;
struct RTPHeader;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
// Callback class used for sending data ready to be packetized
class AudioPacketizationCallback {
public: