Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h |
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h |
index 6006c68f5c6482871f970e125c0ab7b88f0e4a85..a624b23067fef11ca4eded8732f2d30267f2bd91 100644 |
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h |
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h |
@@ -11,6 +11,7 @@ |
#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
+#include <string> |
#include <vector> |
#include "webrtc/base/buffer.h" |
@@ -123,7 +124,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule { |
int RegisterExternalReceiveCodec(int rtp_payload_type, |
AudioDecoder* external_decoder, |
int sample_rate_hz, |
- int num_channels) override; |
+ int num_channels, |
+ const std::string& name) override; |
// Get current received codec. |
int ReceiveCodec(CodecInst* current_codec) const override; |