Moving test-only code (stream_generator) out of vcm implemention.

R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4740 4adac7df-926f-26a2-2b94-8c16560cd09d

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',

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"
 

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"

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 <string.h>
 
@@ -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) /
+    const int packet_size =
-        num_media_packets;
+        (kFrameSize + num_media_packets / 2) / num_media_packets;
     bool marker_bit = (i == num_media_packets - 1);
-    packets_.push_back(GeneratePacket(sequence_number_,
+    packets_.push_back(GeneratePacket(
-                                      timestamp_,
+        sequence_number_, timestamp_, packet_size, (i == 0), marker_bit, type));
-                                      packet_size,
-                                      (i == 0),
-                                      marker_bit,
-                                      type));
     ++sequence_number_;
   }
   for (int i = 0; i < num_empty_packets; ++i) {
-    packets_.push_back(GeneratePacket(sequence_number_,
+    packets_.push_back(GeneratePacket(
-                                      timestamp_,
+        sequence_number_, timestamp_, 0, false, false, kFrameEmpty));
-                                      0,
-                                      false,
-                                      false,
-                                      kFrameEmpty));
     ++sequence_number_;
   }
 }
@@ -119,9 +111,7 @@ bool StreamGenerator::NextPacket(VCMPacket* packet) {
   return true;
 }
 
-void StreamGenerator::DropLastPacket() {
+void StreamGenerator::DropLastPacket() { packets_.pop_back(); }
-  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 {
+int StreamGenerator::PacketsRemaining() const { return packets_.size(); }
-  return packets_.size();
-}
 
 std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
   std::list<VCMPacket>::iterator it = packets_.begin();
   for (int i = 0; i < index; ++i) {
     ++it;
-    if (it == packets_.end()) break;
+    if (it == packets_.end())
+      break;
   }
   return it;
 }

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_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_STREAM_GENERATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
 
 #include <string.h>
 
@@ -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) /
+const unsigned int kFrameSize =
-    (kDefaultFrameRate * 8);
+    (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,
+  void GenerateFrame(FrameType type,
-                     int num_empty_packets, int64_t current_time);
+                     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_