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Removed vie_defines.h

Browse Source 
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}
This commit is contained in:
mflodman
2015-11-06 04:33:51 -08:00
committed by Commit bot
parent d812e14ff1
commit c4a1c370aa
12 changed files with 24 additions and 145 deletions
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6
webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
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@ -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 @@ class RtpPacketHistoryTest : public ::testing::Test {
~RtpPacketHistoryTest() {
delete hist_;
}
SimulatedClock fake_clock_;
RTPPacketHistory* hist_;
enum {kPayload = 127};
@ -54,7 +53,7 @@ class RtpPacketHistoryTest : public ::testing::Test {
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, DynamicExpansion) {
}
TEST_F(RtpPacketHistoryTest, FullExpansion) {
static const int kSendSidePacketHistorySize = 600;
hist_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
size_t len;
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();

4
webrtc/test/channel_transport/channel_transport.cc
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@ -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 @@ void VoiceChannelTransport::IncomingRTCPPacket(
}
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;
}

1
webrtc/video/BUILD.gn
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@ -24,7 +24,6 @@ source_set("video") {
"../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",

1
webrtc/video/video_capture_input.h
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@ -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 {

10
webrtc/video/video_send_stream.cc
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@ -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 @@ int64_t VideoSendStream::GetRtt() const {
}
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 @@ bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
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();

1
webrtc/video/webrtc_video.gypi
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@ -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',

15
webrtc/video_engine/vie_channel.cc
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@ -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 @@ ViEChannel::ViEChannel(uint32_t number_of_cores,
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 @@ ViEChannel::ViEChannel(uint32_t number_of_cores,
}
int32_t ViEChannel::Init() {
static const int kDefaultRenderDelayMs = 10;
module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
// RTP/RTCP initialization.
@ -160,7 +163,7 @@ int32_t ViEChannel::Init() {
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 @@ int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
}
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_)

1
webrtc/video_engine/vie_channel.h
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@ -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"

120
webrtc/video_engine/vie_defines.h
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@ -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_

6
webrtc/video_engine/vie_encoder.cc
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@ -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 int kMinPacingDelayMs = 200;
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 @@ void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
}
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;

1
webrtc/video_engine/vie_encoder.h
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@ -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 {

3
webrtc/video_engine/vie_receiver.h
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@ -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 @@ class ViEReceiver : public RtpData {
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_;