commit f693bfae5ff0a031b4ceeed13655527b411ce650
author Fredrik Solenberg <solenberg@webrtc.org> Tue Dec 11 12:22:10 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Dec 17 10:33:55 2018 +0000
tree 899f0cd708c381a101c090fe1d3c9496e6a85e77
parent a134204aa3ab2b09fd109c59bc3472c92d9e0ae6

Remove CodecInst pt.2

The following APIs on AudioCodingModule are deprecated with this CL:
  static int NumberOfCodecs();
  static int Codec(int, CodecInst*);
  static int Codec(const char*, CodecInst*, int, size_t);
  static int Codec(const char*, int, size_t);
  absl::optional<CodecInst> SendCodec() const;
  bool RegisterReceiveCodec(int, const SdpAudioFormat&);
  int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
  int UnregisterReceiveCodec(uint8_t);
  int32_t ReceiveCodec(CodecInst*);
  absl::optional<SdpAudioFormat> ReceiveFormat();

As well as this method on RtpRtcp module:
  int32_t RegisterSendPayload(const CodecInst&);

Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}

audio/channel_receive.h

diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 9027623..6bbf990 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -13,6 +13,7 @@
 
 #include <map>
 #include <memory>
+#include <utility>
 #include <vector>
 
 #include "absl/types/optional.h"
@@ -79,7 +80,9 @@
   virtual void StartPlayout() = 0;
   virtual void StopPlayout() = 0;
 
-  virtual bool GetRecCodec(CodecInst* codec) const = 0;
+  // Payload type and format of last received RTP packet, if any.
+  virtual absl::optional<std::pair<int, SdpAudioFormat>>
+      GetReceiveCodec() const = 0;
 
   virtual bool ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;