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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #include "webrtc/modules/audio_coding/main/acm2/acm_dump.h" | 11 #include "webrtc/modules/audio_coding/main/acm2/acm_dump.h" |
| 12 | 12 |
| 13 #include <deque> | 13 #include <deque> |
| 14 | 14 |
| 15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
| 16 #include "webrtc/base/thread_annotations.h" | 16 #include "webrtc/base/thread_annotations.h" |
| 17 #include "webrtc/system_wrappers/interface/clock.h" | 17 #include "webrtc/system_wrappers/interface/clock.h" |
| 18 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 18 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 19 #include "webrtc/system_wrappers/interface/file_wrapper.h" | 19 #include "webrtc/system_wrappers/interface/file_wrapper.h" |
| 20 | 20 |
| 21 #ifdef RTC_AUDIOCODING_DEBUG_DUMP | |
| 21 // Files generated at build-time by the protobuf compiler. | 22 // Files generated at build-time by the protobuf compiler. |
| 22 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 23 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 23 #include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" | 24 #include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" |
| 24 #else | 25 #else |
| 25 #include "webrtc/audio_coding/dump.pb.h" | 26 #include "webrtc/audio_coding/dump.pb.h" |
| 26 #endif | 27 #endif |
| 28 #endif | |
| 27 | 29 |
| 28 namespace webrtc { | 30 namespace webrtc { |
| 29 | 31 |
| 30 // Noop implementation if flag is not set | 32 // Noop implementation if flag is not set |
| 31 #ifndef RTC_AUDIOCODING_DEBUG_DUMP | 33 #ifndef RTC_AUDIOCODING_DEBUG_DUMP |
| 32 class AcmDumpImpl final : public AcmDump { | 34 class AcmDumpImpl final : public AcmDump { |
| 33 public: | 35 public: |
| 34 void StartLogging(const std::string& file_name, int duration_ms) override{}; | 36 void StartLogging(const std::string& file_name, int duration_ms) override{}; |
| 35 void LogRtpPacket(bool incoming, | 37 void LogRtpPacket(bool incoming, |
| 36 const uint8_t* packet, | 38 const uint8_t* packet, |
| (...skipping 169 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 206 } | 208 } |
| 207 | 209 |
| 208 void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) { | 210 void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) { |
| 209 recent_log_events_.push_back(event); | 211 recent_log_events_.push_back(event); |
| 210 while (recent_log_events_.front().timestamp_us() < | 212 while (recent_log_events_.front().timestamp_us() < |
| 211 event.timestamp_us() - recent_log_duration_us) { | 213 event.timestamp_us() - recent_log_duration_us) { |
| 212 recent_log_events_.pop_front(); | 214 recent_log_events_.pop_front(); |
| 213 } | 215 } |
| 214 } | 216 } |
| 215 | 217 |
| 216 #endif // RTC_AUDIOCODING_DEBUG_DUMP | |
| 217 | |
| 218 // AcmDump member functions. | |
| 219 rtc::scoped_ptr<AcmDump> AcmDump::Create() { | |
| 220 return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl()); | |
| 221 } | |
| 222 | |
| 223 bool AcmDump::ParseAcmDump(const std::string& file_name, | 218 bool AcmDump::ParseAcmDump(const std::string& file_name, |
|
hlundin-webrtc
2015/07/03 15:13:26
AcmDump::ParseAcmDump() no longer has an implement
| |
| 224 ACMDumpEventStream* result) { | 219 ACMDumpEventStream* result) { |
| 225 char tmp_buffer[1024]; | 220 char tmp_buffer[1024]; |
| 226 int bytes_read = 0; | 221 int bytes_read = 0; |
| 227 rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create()); | 222 rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create()); |
| 228 if (dump_file->OpenFile(file_name.c_str(), true) != 0) { | 223 if (dump_file->OpenFile(file_name.c_str(), true) != 0) { |
| 229 return false; | 224 return false; |
| 230 } | 225 } |
| 231 std::string dump_buffer; | 226 std::string dump_buffer; |
| 232 while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { | 227 while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { |
| 233 dump_buffer.append(tmp_buffer, bytes_read); | 228 dump_buffer.append(tmp_buffer, bytes_read); |
| 234 } | 229 } |
| 235 dump_file->CloseFile(); | 230 dump_file->CloseFile(); |
| 236 return result->ParseFromString(dump_buffer); | 231 return result->ParseFromString(dump_buffer); |
| 237 } | 232 } |
| 238 | 233 |
| 234 #endif // RTC_AUDIOCODING_DEBUG_DUMP | |
| 235 | |
| 236 // AcmDump member functions. | |
| 237 rtc::scoped_ptr<AcmDump> AcmDump::Create() { | |
| 238 return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl()); | |
| 239 } | |
| 239 } // namespace webrtc | 240 } // namespace webrtc |
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