Removed vie_defines.h
The defines still in use was only used in single files, so they were
moved to these specific cc-files.
Review URL: https://codereview.webrtc.org/1411573007
Cr-Commit-Position: refs/heads/master@{#10539}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index d39991e..a406d8b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video_engine/vie_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -29,7 +28,7 @@
~RtpPacketHistoryTest() {
delete hist_;
}
-
+
SimulatedClock fake_clock_;
RTPPacketHistory* hist_;
enum {kPayload = 127};
@@ -54,7 +53,7 @@
array[(*cur_pos)++] = ssrc >> 16;
array[(*cur_pos)++] = ssrc >> 8;
array[(*cur_pos)++] = ssrc;
- }
+ }
};
TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
@@ -268,6 +267,7 @@
}
TEST_F(RtpPacketHistoryTest, FullExpansion) {
+ static const int kSendSidePacketHistorySize = 600;
hist_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
size_t len;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
diff --git a/webrtc/test/channel_transport/channel_transport.cc b/webrtc/test/channel_transport/channel_transport.cc
index 25eb59d..a8aca35 100644
--- a/webrtc/test/channel_transport/channel_transport.cc
+++ b/webrtc/test/channel_transport/channel_transport.cc
@@ -16,7 +16,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#endif
#include "webrtc/test/channel_transport/udp_transport.h"
-#include "webrtc/video_engine/vie_defines.h"
#include "webrtc/voice_engine/include/voe_network.h"
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
@@ -66,10 +65,11 @@
}
int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) {
+ static const int kNumReceiveSocketBuffers = 500;
int return_value = socket_transport_->InitializeReceiveSockets(this,
rtp_port);
if (return_value == 0) {
- return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers);
+ return socket_transport_->StartReceiving(kNumReceiveSocketBuffers);
}
return return_value;
}
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 408bb36..9db0dfc 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -24,7 +24,6 @@
"../video_engine/stream_synchronization.h",
"../video_engine/vie_channel.cc",
"../video_engine/vie_channel.h",
- "../video_engine/vie_defines.h",
"../video_engine/vie_encoder.cc",
"../video_engine/vie_encoder.h",
"../video_engine/vie_receiver.cc",
diff --git a/webrtc/video/video_capture_input.h b/webrtc/video/video_capture_input.h
index 5a86ad2..1affd8d 100644
--- a/webrtc/video/video_capture_input.h
+++ b/webrtc/video/video_capture_input.h
@@ -25,7 +25,6 @@
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/vie_defines.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index a03eacb..958aa33 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -26,7 +26,6 @@
#include "webrtc/video_engine/encoder_state_feedback.h"
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_defines.h"
#include "webrtc/video_engine/vie_encoder.h"
#include "webrtc/video_send_stream.h"
@@ -528,6 +527,7 @@
}
bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
+ static const int kEncoderMinBitrate = 30;
if (video_codec.maxBitrate == 0) {
// Unset max bitrate -> cap to one bit per pixel.
video_codec.maxBitrate =
@@ -535,10 +535,10 @@
1000;
}
- if (video_codec.minBitrate < kViEMinCodecBitrate)
- video_codec.minBitrate = kViEMinCodecBitrate;
- if (video_codec.maxBitrate < kViEMinCodecBitrate)
- video_codec.maxBitrate = kViEMinCodecBitrate;
+ if (video_codec.minBitrate < kEncoderMinBitrate)
+ video_codec.minBitrate = kEncoderMinBitrate;
+ if (video_codec.maxBitrate < kEncoderMinBitrate)
+ video_codec.maxBitrate = kEncoderMinBitrate;
// Stop the media flow while reconfiguring.
vie_encoder_->Pause();
diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi
index f9dbbce..87876e3 100644
--- a/webrtc/video/webrtc_video.gypi
+++ b/webrtc/video/webrtc_video.gypi
@@ -52,7 +52,6 @@
'video_engine/stream_synchronization.h',
'video_engine/vie_channel.cc',
'video_engine/vie_channel.h',
- 'video_engine/vie_defines.h',
'video_engine/vie_encoder.cc',
'video_engine/vie_encoder.h',
'video_engine/vie_receiver.cc',
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index b4e8332..2ffcf8a 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -34,12 +34,14 @@
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/video_engine/report_block_stats.h"
-#include "webrtc/video_engine/vie_defines.h"
namespace webrtc {
const int kMaxDecodeWaitTimeMs = 50;
static const int kMaxTargetDelayMs = 10000;
+const int kMinSendSidePacketHistorySize = 600;
+const int kMaxPacketAgeToNack = 450;
+const int kMaxNackListSize = 250;
// Helper class receiving statistics callbacks.
class ChannelStatsObserver : public CallStatsObserver {
@@ -108,7 +110,7 @@
packet_router_(packet_router),
bandwidth_observer_(bandwidth_observer),
transport_feedback_observer_(transport_feedback_observer),
- nack_history_size_sender_(kSendSidePacketHistorySize),
+ nack_history_size_sender_(kMinSendSidePacketHistorySize),
max_nack_reordering_threshold_(kMaxPacketAgeToNack),
pre_render_callback_(NULL),
report_block_stats_sender_(new ReportBlockStats()),
@@ -138,6 +140,7 @@
}
int32_t ViEChannel::Init() {
+ static const int kDefaultRenderDelayMs = 10;
module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
// RTP/RTCP initialization.
@@ -160,7 +163,7 @@
vcm_->RegisterFrameTypeCallback(this);
vcm_->RegisterReceiveStatisticsCallback(this);
vcm_->RegisterDecoderTimingCallback(this);
- vcm_->SetRenderDelay(kViEDefaultRenderDelayMs);
+ vcm_->SetRenderDelay(kDefaultRenderDelayMs);
module_process_thread_->RegisterModule(vcm_);
module_process_thread_->RegisterModule(&vie_sync_);
@@ -561,12 +564,12 @@
}
if (target_delay_ms == 0) {
// Real-time mode.
- nack_history_size_sender_ = kSendSidePacketHistorySize;
+ nack_history_size_sender_ = kMinSendSidePacketHistorySize;
} else {
nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
// Don't allow a number lower than the default value.
- if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
- nack_history_size_sender_ = kSendSidePacketHistorySize;
+ if (nack_history_size_sender_ < kMinSendSidePacketHistorySize) {
+ nack_history_size_sender_ = kMinSendSidePacketHistorySize;
}
}
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 20eae7d..2f4ba8d 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -22,7 +22,6 @@
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/vie_defines.h"
#include "webrtc/video_engine/vie_receiver.h"
#include "webrtc/video_engine/vie_sync_module.h"
diff --git a/webrtc/video_engine/vie_defines.h b/webrtc/video_engine/vie_defines.h
deleted file mode 100644
index 59b56a5..0000000
--- a/webrtc/video_engine/vie_defines.h
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
-
-#include "webrtc/engine_configurations.h"
-
-// TODO(mflodman) Remove.
-#ifdef WEBRTC_ANDROID
-#include <arpa/inet.h> // NOLINT
-#include <linux/net.h> // NOLINT
-#include <netinet/in.h> // NOLINT
-#include <pthread.h> // NOLINT
-#include <stdio.h> // NOLINT
-#include <stdlib.h> // NOLINT
-#include <string.h> // NOLINT
-#include <sys/socket.h> // NOLINT
-#include <sys/time.h> // NOLINT
-#include <sys/types.h> // NOLINT
-#include <time.h> // NOLINT
-#endif
-
-namespace webrtc {
-
-// General
-enum { kViEMinKeyRequestIntervalMs = 300 };
-
-// ViEBase
-enum { kViEMaxNumberOfChannels = 64 };
-
-// ViECodec
-enum { kViEMaxCodecWidth = 4096 };
-enum { kViEMaxCodecHeight = 3072 };
-enum { kViEMaxCodecFramerate = 60 };
-enum { kViEMinCodecBitrate = 30 };
-
-// ViENetwork
-enum { kViEMaxMtu = 1500 };
-enum { kViESocketThreads = 1 };
-enum { kViENumReceiveSocketBuffers = 500 };
-
-// ViERender
-// Max valid time set in SetRenderTimeoutImage
-enum { kViEMaxRenderTimeoutTimeMs = 10000 };
-// Min valid time set in SetRenderTimeoutImage
-enum { kViEMinRenderTimeoutTimeMs = 33 };
-enum { kViEDefaultRenderDelayMs = 10 };
-
-// ViERTP_RTCP
-enum { kSendSidePacketHistorySize = 600 };
-
-// NACK
-enum { kMaxPacketAgeToNack = 450 }; // In sequence numbers.
-enum { kMaxNackListSize = 250 };
-
-// Id definitions
-enum {
- kViEChannelIdBase = 0x0,
- kViEChannelIdMax = 0xFF,
- kViEDummyChannelId = 0xFFFF
-};
-
-// Module id
-// Create a unique id based on the ViE instance id and the
-// channel id. ViE id > 0 and 0 <= channel id <= 255
-
-inline int ViEId(const int vieId, const int channelId = -1) {
- if (channelId == -1) {
- return static_cast<int>((vieId << 16) + kViEDummyChannelId);
- }
- return static_cast<int>((vieId << 16) + channelId);
-}
-
-inline int ViEModuleId(const int vieId, const int channelId = -1) {
- if (channelId == -1) {
- return static_cast<int>((vieId << 16) + kViEDummyChannelId);
- }
- return static_cast<int>((vieId << 16) + channelId);
-}
-
-inline int ChannelId(const int moduleId) {
- return static_cast<int>(moduleId & 0xffff);
-}
-
-// Windows specific.
-#if defined(_WIN32)
- #define RENDER_MODULE_TYPE kRenderWindows
-
- // Include libraries.
- #pragma comment(lib, "winmm.lib")
-
- #ifndef WEBRTC_EXTERNAL_TRANSPORT
- #pragma comment(lib, "ws2_32.lib")
- #pragma comment(lib, "Iphlpapi.lib") // _GetAdaptersAddresses
- #endif
-#endif
-
-// Mac specific.
-#ifdef WEBRTC_MAC
- #define SLEEP(x) usleep(x * 1000)
- #define RENDER_MODULE_TYPE kRenderWindows
-#endif
-
-// Android specific.
-#ifdef WEBRTC_ANDROID
- #define FAR
- #define __cdecl
-#endif // WEBRTC_ANDROID
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 96ed2f9..fad5843 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -33,7 +33,6 @@
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video/send_statistics_proxy.h"
#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/vie_defines.h"
namespace webrtc {
@@ -46,6 +45,8 @@
static const float kStopPaddingThresholdMs = 2000;
+static const int kMinKeyFrameRequestIntervalMs = 300;
+
std::vector<uint32_t> AllocateStreamBitrates(
uint32_t total_bitrate,
const SimulcastStream* stream_configs,
@@ -587,7 +588,8 @@
}
int64_t now = TickTime::MillisecondTimestamp();
- if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
+ if (time_last_intra_request_ms_[ssrc] + kMinKeyFrameRequestIntervalMs
+ > now) {
return;
}
time_last_intra_request_ms_[ssrc] = now;
diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h
index 872fbb6..66ac4a5 100644
--- a/webrtc/video_engine/vie_encoder.h
+++ b/webrtc/video_engine/vie_encoder.h
@@ -25,7 +25,6 @@
#include "webrtc/modules/video_processing/main/interface/video_processing.h"
#include "webrtc/typedefs.h"
#include "webrtc/video/video_capture_input.h"
-#include "webrtc/video_engine/vie_defines.h"
namespace webrtc {
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index 20a9627..d75622f 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -18,7 +18,6 @@
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/vie_defines.h"
namespace webrtc {
@@ -118,7 +117,7 @@
rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_;
bool receiving_;
- uint8_t restored_packet_[kViEMaxMtu];
+ uint8_t restored_packet_[IP_PACKET_SIZE];
bool restored_packet_in_use_;
bool receiving_ast_enabled_;
bool receiving_cvo_enabled_;
