Chromium Code Reviews| Index: webrtc/modules/audio_coding/main/acm2/acm_dump.cc |
| diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc |
| index 3fc055723aa0a9fbdf0126807ba10cab9e99ed65..9c624d97b67eb728c6f54729ee8a37e6359f2631 100644 |
| --- a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc |
| +++ b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc |
| @@ -18,12 +18,14 @@ |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/file_wrapper.h" |
| +#ifdef RTC_AUDIOCODING_DEBUG_DUMP |
| // Files generated at build-time by the protobuf compiler. |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" |
| #else |
| #include "webrtc/audio_coding/dump.pb.h" |
| #endif |
| +#endif |
| namespace webrtc { |
| @@ -213,13 +215,6 @@ void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) { |
| } |
| } |
| -#endif // RTC_AUDIOCODING_DEBUG_DUMP |
| - |
| -// AcmDump member functions. |
| -rtc::scoped_ptr<AcmDump> AcmDump::Create() { |
| - return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl()); |
| -} |
| - |
| bool AcmDump::ParseAcmDump(const std::string& file_name, |
|
hlundin-webrtc
2015/07/03 15:13:26
AcmDump::ParseAcmDump() no longer has an implement
|
| ACMDumpEventStream* result) { |
| char tmp_buffer[1024]; |
| @@ -236,4 +231,10 @@ bool AcmDump::ParseAcmDump(const std::string& file_name, |
| return result->ParseFromString(dump_buffer); |
| } |
| +#endif // RTC_AUDIOCODING_DEBUG_DUMP |
| + |
| +// AcmDump member functions. |
| +rtc::scoped_ptr<AcmDump> AcmDump::Create() { |
| + return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl()); |
| +} |
| } // namespace webrtc |