commit 84f75699c65384a961cdf9ebffefd6bc83e0d8e2
author Henrik Lundin <henrik.lundin@webrtc.org> Wed Feb 01 12:07:10 2023 +0000
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Feb 01 16:09:26 2023 +0000
tree ea5761e014beca19fe25b7e13842bcd2baead6a4
parent c5455e7b5391ec4b567f7dd0bdb88bdc02fdf07c

Break apart AudioCodingModule and AcmReceiver

This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.

The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.

Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}

modules/audio_coding/test/EncodeDecodeTest.h

diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 89b7644..9cd2c23 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -15,6 +15,7 @@
 #include <string.h>
 
 #include "absl/strings/string_view.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/test/PCMFile.h"
 #include "modules/audio_coding/test/RTPFile.h"
@@ -73,7 +74,7 @@
  public:
   Receiver();
   virtual ~Receiver() {}
-  void Setup(AudioCodingModule* acm,
+  void Setup(acm2::AcmReceiver* acm_receiver,
              RTPStream* rtpStream,
              absl::string_view out_file_name,
              size_t channels,
@@ -91,7 +92,7 @@
   bool _firstTime;
 
  protected:
-  AudioCodingModule* _acm;
+  acm2::AcmReceiver* _acm_receiver;
   uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
   RTPStream* _rtpStream;
   RTPHeader _rtpHeader;