diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 6aa100ef37..c1de3b479a 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -204,12 +204,12 @@ 'video_coding/main/source/jitter_buffer_unittest.cc', 'video_coding/main/source/receiver_unittest.cc', 'video_coding/main/source/session_info_unittest.cc', - 'video_coding/main/source/stream_generator.cc', - 'video_coding/main/source/stream_generator.h', 'video_coding/main/source/timing_unittest.cc', 'video_coding/main/source/video_coding_robustness_unittest.cc', 'video_coding/main/source/video_coding_impl_unittest.cc', 'video_coding/main/source/qm_select_unittest.cc', + 'video_coding/main/source/test/stream_generator.cc', + 'video_coding/main/source/test/stream_generator.h', 'video_coding/main/test/pcap_file_reader.cc', 'video_coding/main/test/pcap_file_reader_unittest.cc', 'video_coding/main/test/rtp_file_reader.cc', diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc index 29e5250d88..f596d68002 100644 --- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc +++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc @@ -17,7 +17,7 @@ #include "webrtc/modules/video_coding/main/source/jitter_buffer.h" #include "webrtc/modules/video_coding/main/source/media_opt_util.h" #include "webrtc/modules/video_coding/main/source/packet.h" -#include "webrtc/modules/video_coding/main/source/stream_generator.h" +#include "webrtc/modules/video_coding/main/source/test/stream_generator.h" #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/system_wrappers/interface/clock.h" diff --git a/webrtc/modules/video_coding/main/source/receiver_unittest.cc b/webrtc/modules/video_coding/main/source/receiver_unittest.cc index 7441baecc3..33a3d95f96 100644 --- a/webrtc/modules/video_coding/main/source/receiver_unittest.cc +++ b/webrtc/modules/video_coding/main/source/receiver_unittest.cc @@ -14,7 +14,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/modules/video_coding/main/source/receiver.h" -#include "webrtc/modules/video_coding/main/source/stream_generator.h" +#include "webrtc/modules/video_coding/main/source/test/stream_generator.h" #include "webrtc/modules/video_coding/main/source/timing.h" #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/system_wrappers/interface/clock.h" diff --git a/webrtc/modules/video_coding/main/source/stream_generator.cc b/webrtc/modules/video_coding/main/source/test/stream_generator.cc similarity index 78% rename from webrtc/modules/video_coding/main/source/stream_generator.cc rename to webrtc/modules/video_coding/main/source/test/stream_generator.cc index 652b014061..4f85dffc50 100644 --- a/webrtc/modules/video_coding/main/source/stream_generator.cc +++ b/webrtc/modules/video_coding/main/source/test/stream_generator.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/video_coding/main/source/stream_generator.h" +#include "webrtc/modules/video_coding/main/source/test/stream_generator.h" #include @@ -19,7 +19,6 @@ #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/system_wrappers/interface/clock.h" - namespace webrtc { StreamGenerator::StreamGenerator(uint16_t start_seq_num, @@ -30,7 +29,8 @@ StreamGenerator::StreamGenerator(uint16_t start_seq_num, timestamp_(start_timestamp), start_time_(current_time) {} -void StreamGenerator::Init(uint16_t start_seq_num, uint32_t start_timestamp, +void StreamGenerator::Init(uint16_t start_seq_num, + uint32_t start_timestamp, int64_t current_time) { packets_.clear(); sequence_number_ = start_seq_num; @@ -45,24 +45,16 @@ void StreamGenerator::GenerateFrame(FrameType type, int64_t current_time) { timestamp_ = 90 * (current_time - start_time_); for (int i = 0; i < num_media_packets; ++i) { - const int packet_size = (kFrameSize + num_media_packets / 2) / - num_media_packets; + const int packet_size = + (kFrameSize + num_media_packets / 2) / num_media_packets; bool marker_bit = (i == num_media_packets - 1); - packets_.push_back(GeneratePacket(sequence_number_, - timestamp_, - packet_size, - (i == 0), - marker_bit, - type)); + packets_.push_back(GeneratePacket( + sequence_number_, timestamp_, packet_size, (i == 0), marker_bit, type)); ++sequence_number_; } for (int i = 0; i < num_empty_packets; ++i) { - packets_.push_back(GeneratePacket(sequence_number_, - timestamp_, - 0, - false, - false, - kFrameEmpty)); + packets_.push_back(GeneratePacket( + sequence_number_, timestamp_, 0, false, false, kFrameEmpty)); ++sequence_number_; } } @@ -119,9 +111,7 @@ bool StreamGenerator::NextPacket(VCMPacket* packet) { return true; } -void StreamGenerator::DropLastPacket() { - packets_.pop_back(); -} +void StreamGenerator::DropLastPacket() { packets_.pop_back(); } uint16_t StreamGenerator::NextSequenceNumber() const { if (packets_.empty()) @@ -129,15 +119,14 @@ uint16_t StreamGenerator::NextSequenceNumber() const { return packets_.front().seqNum; } -int StreamGenerator::PacketsRemaining() const { - return packets_.size(); -} +int StreamGenerator::PacketsRemaining() const { return packets_.size(); } std::list::iterator StreamGenerator::GetPacketIterator(int index) { std::list::iterator it = packets_.begin(); for (int i = 0; i < index; ++i) { ++it; - if (it == packets_.end()) break; + if (it == packets_.end()) + break; } return it; } diff --git a/webrtc/modules/video_coding/main/source/stream_generator.h b/webrtc/modules/video_coding/main/source/test/stream_generator.h similarity index 72% rename from webrtc/modules/video_coding/main/source/stream_generator.h rename to webrtc/modules/video_coding/main/source/test/stream_generator.h index dd6deadaad..6565527db6 100644 --- a/webrtc/modules/video_coding/main/source/stream_generator.h +++ b/webrtc/modules/video_coding/main/source/test/stream_generator.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_STREAM_GENERATOR_H_ -#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_STREAM_GENERATOR_H_ +#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_ +#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_ #include @@ -22,21 +22,23 @@ namespace webrtc { const unsigned int kDefaultBitrateKbps = 1000; const unsigned int kDefaultFrameRate = 25; const unsigned int kMaxPacketSize = 1500; -const unsigned int kFrameSize = (kDefaultBitrateKbps + kDefaultFrameRate * 4) / - (kDefaultFrameRate * 8); +const unsigned int kFrameSize = + (kDefaultBitrateKbps + kDefaultFrameRate * 4) / (kDefaultFrameRate * 8); const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate; - - class StreamGenerator { public: - StreamGenerator(uint16_t start_seq_num, uint32_t start_timestamp, + StreamGenerator(uint16_t start_seq_num, + uint32_t start_timestamp, int64_t current_time); - void Init(uint16_t start_seq_num, uint32_t start_timestamp, + void Init(uint16_t start_seq_num, + uint32_t start_timestamp, int64_t current_time); - void GenerateFrame(FrameType type, int num_media_packets, - int num_empty_packets, int64_t current_time); + void GenerateFrame(FrameType type, + int num_media_packets, + int num_empty_packets, + int64_t current_time); VCMPacket GeneratePacket(uint16_t sequence_number, uint32_t timestamp, @@ -70,4 +72,4 @@ class StreamGenerator { } // namespace webrtc -#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_STREAM_GENERATOR_H_ +#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_