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Unified Diff: webrtc/modules/audio_coding/include/audio_coding_module.h

Issue 1484343003: NetEq: Add codec name and RTP timestamp rate to DecoderInfo (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years ago
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Index: webrtc/modules/audio_coding/include/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 844bd57cd19a81d8f516cc361e519f26adf6531b..f5af65a13d5d12b84ecf86c992175d821eb402cc 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#include <string>
#include <vector>
#include "webrtc/base/optional.h"
@@ -471,10 +472,14 @@ class AudioCodingModule {
//
virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
+ // Registers an external decoder. The name is only used to provide information
+ // back to the caller about the decoder. Hence, the name is arbitrary, and may
+ // be empty.
virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
AudioDecoder* external_decoder,
int sample_rate_hz,
- int num_channels) = 0;
+ int num_channels,
+ const std::string& name) = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t UnregisterReceiveCodec()

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