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Remove rtp_rtcp/ dump functionality.

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Removes RTP dumping from VideoEngine and VoiceEngine.

BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47179004

Cr-Commit-Position: refs/heads/master@{#9234}
This commit is contained in:
Peter Boström
2015-05-20 14:11:13 +02:00
parent ca667dbfdd
commit c3f4dbc40b
19 changed files with 8 additions and 744 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
talk/media/webrtc/fakewebrtcvoiceengine.h
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@@ -1035,10 +1035,6 @@ class FakeWebRtcVoiceEngine
channels_[channel]->nack_max_packets = maxNoPackets; channels_[channel]->nack_max_packets = maxNoPackets;
return 0; return 0;
} }
WEBRTC_STUB(StartRTPDump, (int channel, const char* fileNameUTF8,
webrtc::RTPDirections direction));
WEBRTC_STUB(StopRTPDump, (int channel, webrtc::RTPDirections direction));
WEBRTC_STUB(RTPDumpIsActive, (int channel, webrtc::RTPDirections direction));
WEBRTC_STUB(InsertExtraRTPPacket, (int channel, unsigned char payloadType, WEBRTC_STUB(InsertExtraRTPPacket, (int channel, unsigned char payloadType,
bool markerBit, const char* payloadData, bool markerBit, const char* payloadData,
unsigned short payloadSize)); unsigned short payloadSize));

16
webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
View File

@@ -400,22 +400,6 @@ JOWW(jint, VoiceEngine_stopDebugRecording)(JNIEnv* jni, jobject j_voe) {
return voe_data->apm->StopDebugRecording(); return voe_data->apm->StopDebugRecording();
} }
JOWW(jint, VoiceEngine_startRtpDump)(JNIEnv* jni, jobject j_voe, jint channel,
jstring j_filename, jint direction) {
VoiceEngineData* voe_data = GetVoiceEngineData(jni, j_voe);
std::string filename = JavaToStdString(jni, j_filename);
return voe_data->rtp->StartRTPDump(
channel, filename.c_str(),
static_cast<webrtc::RTPDirections>(direction));
}
JOWW(jint, VoiceEngine_stopRtpDump)(JNIEnv* jni, jobject j_voe, jint channel,
jint direction) {
VoiceEngineData* voe_data = GetVoiceEngineData(jni, j_voe);
return voe_data->rtp->StopRTPDump(
channel, static_cast<webrtc::RTPDirections>(direction));
}
JOWW(void, CodecInst_dispose)(JNIEnv* jni, jobject j_codec) { JOWW(void, CodecInst_dispose)(JNIEnv* jni, jobject j_codec) {
delete GetCodecInst(jni, j_codec); delete GetCodecInst(jni, j_codec);
} }

3
webrtc/modules/utility/BUILD.gn
View File

@@ -16,7 +16,6 @@ source_set("utility") {
"interface/helpers_android.h", "interface/helpers_android.h",
"interface/jvm_android.h", "interface/jvm_android.h",
"interface/process_thread.h", "interface/process_thread.h",
"interface/rtp_dump.h",
"source/audio_frame_operations.cc", "source/audio_frame_operations.cc",
"source/coder.cc", "source/coder.cc",
"source/coder.h", "source/coder.h",
@@ -28,8 +27,6 @@ source_set("utility") {
"source/jvm_android.cc", "source/jvm_android.cc",
"source/process_thread_impl.cc", "source/process_thread_impl.cc",
"source/process_thread_impl.h", "source/process_thread_impl.h",
"source/rtp_dump_impl.cc",
"source/rtp_dump_impl.h",
] ]
configs += [ "../..:common_config" ] configs += [ "../..:common_config" ]

52
webrtc/modules/utility/interface/rtp_dump.h
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@@ -1,52 +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.
*/
// This file implements a class that writes a stream of RTP and RTCP packets
// to a file according to the format specified by rtpplay. See
// http://www.cs.columbia.edu/irt/software/rtptools/.
// Notes: supported platforms are Windows, Linux and Mac OSX
#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
#include "webrtc/system_wrappers/interface/file_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RtpDump
{
public:
// Factory method.
static RtpDump* CreateRtpDump();
// Delete function. Destructor disabled.
static void DestroyRtpDump(RtpDump* object);
// Open the file fileNameUTF8 for writing RTP/RTCP packets.
// Note: this API also adds the rtpplay header.
virtual int32_t Start(const char* fileNameUTF8) = 0;
// Close the existing file. No more packets will be recorded.
virtual int32_t Stop() = 0;
// Return true if a file is open for recording RTP/RTCP packets.
virtual bool IsActive() const = 0;
// Writes the RTP/RTCP packet in packet with length packetLength in bytes.
// Note: packet should contain the RTP/RTCP part of the packet. I.e. the
// first bytes of packet should be the RTP/RTCP header.
virtual int32_t DumpPacket(const uint8_t* packet,
size_t packetLength) = 0;
protected:
virtual ~RtpDump();
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_

249
webrtc/modules/utility/source/rtp_dump_impl.cc
View File

@@ -1,249 +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.
*/
#include "webrtc/modules/utility/source/rtp_dump_impl.h"
#include <assert.h>
#include <stdio.h>
#include <limits>
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#if defined(_WIN32)
#include <Windows.h>
#include <mmsystem.h>
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
#include <string.h>
#include <sys/time.h>
#include <time.h>
#endif
#if (defined(_DEBUG) && defined(_WIN32))
#define DEBUG_PRINT(expr) OutputDebugString(##expr)
#define DEBUG_PRINTP(expr, p) \
{ \
char msg[128]; \
sprintf(msg, ##expr, p); \
OutputDebugString(msg); \
}
#else
#define DEBUG_PRINT(expr) ((void)0)
#define DEBUG_PRINTP(expr,p) ((void)0)
#endif // defined(_DEBUG) && defined(_WIN32)
namespace webrtc {
const char RTPFILE_VERSION[] = "1.0";
const uint32_t MAX_UWORD32 = 0xffffffff;
// This stucture is specified in the rtpdump documentation.
// This struct corresponds to RD_packet_t in
// http://www.cs.columbia.edu/irt/software/rtptools/
typedef struct
{
// Length of packet, including this header (may be smaller than plen if not
// whole packet recorded).
uint16_t length;
// Actual header+payload length for RTP, 0 for RTCP.
uint16_t plen;
// Milliseconds since the start of recording.
uint32_t offset;
} RtpDumpPacketHeader;
RtpDump* RtpDump::CreateRtpDump()
{
return new RtpDumpImpl();
}
void RtpDump::DestroyRtpDump(RtpDump* object)
{
delete object;
}
RtpDumpImpl::RtpDumpImpl()
: _critSect(CriticalSectionWrapper::CreateCriticalSection()),
_file(*FileWrapper::Create()),
_startTime(0)
{
}
RtpDump::~RtpDump()
{
}
RtpDumpImpl::~RtpDumpImpl()
{
_file.Flush();
_file.CloseFile();
delete &_file;
delete _critSect;
}
int32_t RtpDumpImpl::Start(const char* fileNameUTF8)
{
if (fileNameUTF8 == NULL)
{
return -1;
}
CriticalSectionScoped lock(_critSect);
_file.Flush();
_file.CloseFile();
if (_file.OpenFile(fileNameUTF8, false, false, false) == -1)
{
LOG(LS_ERROR) << "Failed to open file.";
return -1;
}
// Store start of RTP dump (to be used for offset calculation later).
_startTime = GetTimeInMS();
// All rtp dump files start with #!rtpplay.
char magic[16];
sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION);
if (_file.WriteText(magic) == -1)
{
LOG(LS_ERROR) << "Error writing to file.";
return -1;
}
// The header according to the rtpdump documentation is sizeof(RD_hdr_t)
// which is 8 + 4 + 2 = 14 bytes for 32-bit architecture (and 22 bytes on
// 64-bit architecture). However, Wireshark use 16 bytes for the header
// regardless of if the binary is 32-bit or 64-bit. Go by the same approach
// as Wireshark since it makes more sense.
// http://wiki.wireshark.org/rtpdump explains that an additional 2 bytes
// of padding should be added to the header.
char dummyHdr[16];
memset(dummyHdr, 0, 16);
if (!_file.Write(dummyHdr, sizeof(dummyHdr)))
{
LOG(LS_ERROR) << "Error writing to file.";
return -1;
}
return 0;
}
int32_t RtpDumpImpl::Stop()
{
CriticalSectionScoped lock(_critSect);
_file.Flush();
_file.CloseFile();
return 0;
}
bool RtpDumpImpl::IsActive() const
{
CriticalSectionScoped lock(_critSect);
return _file.Open();
}
int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, size_t packetLength)
{
CriticalSectionScoped lock(_critSect);
if (!IsActive())
{
return 0;
}
if (packet == NULL)
{
return -1;
}
RtpDumpPacketHeader hdr;
size_t total_size = packetLength + sizeof hdr;
if (packetLength < 1 || total_size > std::numeric_limits<uint16_t>::max())
{
return -1;
}
// If the packet doesn't contain a valid RTCP header the packet will be
// considered RTP (without further verification).
bool isRTCP = RTCP(packet);
// Offset is relative to when recording was started.
uint32_t offset = GetTimeInMS();
if (offset < _startTime)
{
// Compensate for wraparound.
offset += MAX_UWORD32 - _startTime + 1;
} else {
offset -= _startTime;
}
hdr.offset = RtpDumpHtonl(offset);
hdr.length = RtpDumpHtons((uint16_t)(total_size));
if (isRTCP)
{
hdr.plen = 0;
}
else
{
hdr.plen = RtpDumpHtons((uint16_t)packetLength);
}
if (!_file.Write(&hdr, sizeof(hdr)))
{
LOG(LS_ERROR) << "Error writing to file.";
return -1;
}
if (!_file.Write(packet, packetLength))
{
LOG(LS_ERROR) << "Error writing to file.";
return -1;
}
return 0;
}
bool RtpDumpImpl::RTCP(const uint8_t* packet) const
{
return packet[1] == 192 || packet[1] == 200 || packet[1] == 201 ||
packet[1] == 202 || packet[1] == 203 || packet[1] == 204 ||
packet[1] == 205 || packet[1] == 206 || packet[1] == 207;
}
// TODO (hellner): why is TickUtil not used here?
inline uint32_t RtpDumpImpl::GetTimeInMS() const
{
#if defined(_WIN32)
return timeGetTime();
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
struct timeval tv;
struct timezone tz;
unsigned long val;
gettimeofday(&tv, &tz);
val = tv.tv_sec * 1000 + tv.tv_usec / 1000;
return val;
#endif
}
inline uint32_t RtpDumpImpl::RtpDumpHtonl(uint32_t x) const
{
#if defined(WEBRTC_ARCH_BIG_ENDIAN)
return x;
#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
return (x >> 24) + ((((x >> 16) & 0xFF) << 8) + ((((x >> 8) & 0xFF) << 16) +
((x & 0xFF) << 24)));
#endif
}
inline uint16_t RtpDumpImpl::RtpDumpHtons(uint16_t x) const
{
#if defined(WEBRTC_ARCH_BIG_ENDIAN)
return x;
#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
return (x >> 8) + ((x & 0xFF) << 8);
#endif
}
} // namespace webrtc

49
webrtc/modules/utility/source/rtp_dump_impl.h
View File

@@ -1,49 +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_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
#include "webrtc/modules/utility/interface/rtp_dump.h"
namespace webrtc {
class CriticalSectionWrapper;
class FileWrapper;
class RtpDumpImpl : public RtpDump
{
public:
RtpDumpImpl();
virtual ~RtpDumpImpl();
int32_t Start(const char* fileNameUTF8) override;
int32_t Stop() override;
bool IsActive() const override;
int32_t DumpPacket(const uint8_t* packet, size_t packetLength) override;
private:
// Return the system time in ms.
inline uint32_t GetTimeInMS() const;
// Return x in network byte order (big endian).
inline uint32_t RtpDumpHtonl(uint32_t x) const;
// Return x in network byte order (big endian).
inline uint16_t RtpDumpHtons(uint16_t x) const;
// Return true if the packet starts with a valid RTCP header.
// Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how
// to determine if the packet is an RTCP packet.
bool RTCP(const uint8_t* packet) const;
private:
CriticalSectionWrapper* _critSect;
FileWrapper& _file;
uint32_t _startTime;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_

3
webrtc/modules/utility/utility.gypi
View File

@@ -24,7 +24,6 @@
'interface/helpers_android.h', 'interface/helpers_android.h',
'interface/jvm_android.h', 'interface/jvm_android.h',
'interface/process_thread.h', 'interface/process_thread.h',
'interface/rtp_dump.h',
'source/audio_frame_operations.cc', 'source/audio_frame_operations.cc',
'source/coder.cc', 'source/coder.cc',
'source/coder.h', 'source/coder.h',
@@ -36,8 +35,6 @@
'source/jvm_android.cc', 'source/jvm_android.cc',
'source/process_thread_impl.cc', 'source/process_thread_impl.cc',
'source/process_thread_impl.h', 'source/process_thread_impl.h',
'source/rtp_dump_impl.cc',
'source/rtp_dump_impl.h',
], ],
}, },
], # targets ], # targets

18
webrtc/video_engine/vie_channel.cc
View File

@@ -105,7 +105,6 @@ ViEChannel::ViEChannel(int32_t channel_id,
nullptr, nullptr,
nullptr)), nullptr)),
vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this), vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
vie_sender_(channel_id),
vie_sync_(vcm_, this), vie_sync_(vcm_, this),
stats_observer_(new ChannelStatsObserver(this)), stats_observer_(new ChannelStatsObserver(this)),
vcm_receive_stats_callback_(NULL), vcm_receive_stats_callback_(NULL),
@@ -1294,23 +1293,6 @@ void ViEChannel::RegisterSendBitrateObserver(
send_bitrate_observer_.Set(observer); send_bitrate_observer_.Set(observer);
} }
int32_t ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
RTPDirections direction) {
if (direction == kRtpIncoming) {
return vie_receiver_.StartRTPDump(file_nameUTF8);
} else {
return vie_sender_.StartRTPDump(file_nameUTF8);
}
}
int32_t ViEChannel::StopRTPDump(RTPDirections direction) {
if (direction == kRtpIncoming) {
return vie_receiver_.StopRTPDump();
} else {
return vie_sender_.StopRTPDump();
}
}
int32_t ViEChannel::StartSend() { int32_t ViEChannel::StartSend() {
CriticalSectionScoped cs(callback_cs_.get()); CriticalSectionScoped cs(callback_cs_.get());
if (!external_transport_) { if (!external_transport_) {

4
webrtc/video_engine/vie_channel.h
View File

@@ -248,10 +248,6 @@ class ViEChannel
// Called on any new send bitrate estimate. // Called on any new send bitrate estimate.
void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer); void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
int32_t StartRTPDump(const char file_nameUTF8[1024],
RTPDirections direction);
int32_t StopRTPDump(RTPDirections direction);
// Implements RtpFeedback. // Implements RtpFeedback.
virtual int32_t OnInitializeDecoder( virtual int32_t OnInitializeDecoder(
const int32_t id, const int32_t id,

47
webrtc/video_engine/vie_receiver.cc
View File

@@ -21,7 +21,6 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/logging.h"
@@ -55,7 +54,6 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
vcm_(module_vcm), vcm_(module_vcm),
remote_bitrate_estimator_(remote_bitrate_estimator), remote_bitrate_estimator_(remote_bitrate_estimator),
ntp_estimator_(new RemoteNtpTimeEstimator(clock_)), ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
rtp_dump_(NULL),
receiving_(false), receiving_(false),
restored_packet_in_use_(false), restored_packet_in_use_(false),
receiving_ast_enabled_(false), receiving_ast_enabled_(false),
@@ -66,11 +64,6 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
ViEReceiver::~ViEReceiver() { ViEReceiver::~ViEReceiver() {
UpdateHistograms(); UpdateHistograms();
if (rtp_dump_) {
rtp_dump_->Stop();
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
}
} }
void ViEReceiver::UpdateHistograms() { void ViEReceiver::UpdateHistograms() {
@@ -254,9 +247,6 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
if (!receiving_) { if (!receiving_) {
return -1; return -1;
} }
if (rtp_dump_) {
rtp_dump_->DumpPacket(rtp_packet, rtp_packet_length);
}
} }
RTPHeader header; RTPHeader header;
@@ -407,10 +397,6 @@ int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
return -1; return -1;
} }
if (rtp_dump_) {
rtp_dump_->DumpPacket(rtcp_packet, rtcp_packet_length);
}
std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin(); std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
while (it != rtp_rtcp_simulcast_.end()) { while (it != rtp_rtcp_simulcast_.end()) {
RtpRtcp* rtp_rtcp = *it++; RtpRtcp* rtp_rtcp = *it++;
@@ -452,39 +438,6 @@ void ViEReceiver::StopReceive() {
receiving_ = false; receiving_ = false;
} }
int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) {
CriticalSectionScoped cs(receive_cs_.get());
if (rtp_dump_) {
// Restart it if it already exists and is started
rtp_dump_->Stop();
} else {
rtp_dump_ = RtpDump::CreateRtpDump();
if (rtp_dump_ == NULL) {
return -1;
}
}
if (rtp_dump_->Start(file_nameUTF8) != 0) {
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
return -1;
}
return 0;
}
int ViEReceiver::StopRTPDump() {
CriticalSectionScoped cs(receive_cs_.get());
if (rtp_dump_) {
if (rtp_dump_->IsActive()) {
rtp_dump_->Stop();
}
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
} else {
return -1;
}
return 0;
}
ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const { ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
return rtp_receive_statistics_.get(); return rtp_receive_statistics_.get();
} }

5
webrtc/video_engine/vie_receiver.h
View File

@@ -27,7 +27,6 @@ class FecReceiver;
class RemoteNtpTimeEstimator; class RemoteNtpTimeEstimator;
class ReceiveStatistics; class ReceiveStatistics;
class RemoteBitrateEstimator; class RemoteBitrateEstimator;
class RtpDump;
class RtpHeaderParser; class RtpHeaderParser;
class RTPPayloadRegistry; class RTPPayloadRegistry;
class RtpReceiver; class RtpReceiver;
@@ -68,9 +67,6 @@ class ViEReceiver : public RtpData {
void StartReceive(); void StartReceive();
void StopReceive(); void StopReceive();
int StartRTPDump(const char file_nameUTF8[1024]);
int StopRTPDump();
// Receives packets from external transport. // Receives packets from external transport.
int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length, int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
const PacketTime& packet_time); const PacketTime& packet_time);
@@ -115,7 +111,6 @@ class ViEReceiver : public RtpData {
rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_; rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_;
RtpDump* rtp_dump_;
bool receiving_; bool receiving_;
uint8_t restored_packet_[kViEMaxMtu]; uint8_t restored_packet_[kViEMaxMtu];
bool restored_packet_in_use_; bool restored_packet_in_use_;

71
webrtc/video_engine/vie_sender.cc
View File

@@ -10,28 +10,15 @@
#include "webrtc/video_engine/vie_sender.h" #include "webrtc/video_engine/vie_sender.h"
#include <assert.h>
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc { namespace webrtc {
ViESender::ViESender(int channel_id) ViESender::ViESender()
: channel_id_(channel_id), : critsect_(CriticalSectionWrapper::CreateCriticalSection()),
critsect_(CriticalSectionWrapper::CreateCriticalSection()), transport_(NULL) {
transport_(NULL),
rtp_dump_(NULL) {
}
ViESender::~ViESender() {
if (rtp_dump_) {
rtp_dump_->Stop();
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
}
} }
int ViESender::RegisterSendTransport(Transport* transport) { int ViESender::RegisterSendTransport(Transport* transport) {
@@ -52,66 +39,22 @@ int ViESender::DeregisterSendTransport() {
return 0; return 0;
} }
int ViESender::StartRTPDump(const char file_nameUTF8[1024]) { int ViESender::SendPacket(int id, const void* data, size_t len) {
CriticalSectionScoped cs(critsect_.get());
if (rtp_dump_) {
// Packet dump is already started, restart it.
rtp_dump_->Stop();
} else {
rtp_dump_ = RtpDump::CreateRtpDump();
if (rtp_dump_ == NULL) {
return -1;
}
}
if (rtp_dump_->Start(file_nameUTF8) != 0) {
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
return -1;
}
return 0;
}
int ViESender::StopRTPDump() {
CriticalSectionScoped cs(critsect_.get());
if (rtp_dump_) {
if (rtp_dump_->IsActive()) {
rtp_dump_->Stop();
}
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
} else {
return -1;
}
return 0;
}
int ViESender::SendPacket(int vie_id, const void* data, size_t len) {
CriticalSectionScoped cs(critsect_.get()); CriticalSectionScoped cs(critsect_.get());
if (!transport_) { if (!transport_) {
// No transport // No transport
return -1; return -1;
} }
assert(ChannelId(vie_id) == channel_id_); return transport_->SendPacket(id, data, len);
if (rtp_dump_) {
rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len);
}
return transport_->SendPacket(channel_id_, data, len);
} }
int ViESender::SendRTCPPacket(int vie_id, const void* data, size_t len) { int ViESender::SendRTCPPacket(int id, const void* data, size_t len) {
CriticalSectionScoped cs(critsect_.get()); CriticalSectionScoped cs(critsect_.get());
if (!transport_) { if (!transport_) {
return -1; return -1;
} }
assert(ChannelId(vie_id) == channel_id_);
if (rtp_dump_) { return transport_->SendRTCPPacket(id, data, len);
rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len);
}
return transport_->SendRTCPPacket(channel_id_, data, len);
} }
} // namespace webrtc } // namespace webrtc

11
webrtc/video_engine/vie_sender.h
View File

@@ -22,34 +22,25 @@
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;
class RtpDump;
class Transport; class Transport;
class VideoCodingModule; class VideoCodingModule;
class ViESender: public Transport { class ViESender: public Transport {
public: public:
explicit ViESender(const int32_t channel_id); ViESender();
~ViESender();
// Registers transport to use for sending RTP and RTCP. // Registers transport to use for sending RTP and RTCP.
int RegisterSendTransport(Transport* transport); int RegisterSendTransport(Transport* transport);
int DeregisterSendTransport(); int DeregisterSendTransport();
// Stores all incoming packets to file.
int StartRTPDump(const char file_nameUTF8[1024]);
int StopRTPDump();
// Implements Transport. // Implements Transport.
int SendPacket(int vie_id, const void* data, size_t len) override; int SendPacket(int vie_id, const void* data, size_t len) override;
int SendRTCPPacket(int vie_id, const void* data, size_t len) override; int SendRTCPPacket(int vie_id, const void* data, size_t len) override;
private: private:
const int32_t channel_id_;
rtc::scoped_ptr<CriticalSectionWrapper> critsect_; rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
Transport* transport_; Transport* transport_;
RtpDump* rtp_dump_;
}; };
} // namespace webrtc } // namespace webrtc

111
webrtc/voice_engine/channel.cc
View File

@@ -25,7 +25,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/utility/interface/audio_frame_operations.h" #include "webrtc/modules/utility/interface/audio_frame_operations.h"
#include "webrtc/modules/utility/interface/process_thread.h" #include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h" #include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/system_wrappers/interface/trace.h"
@@ -245,14 +244,6 @@ Channel::SendPacket(int channel, const void *data, size_t len)
uint8_t* bufferToSendPtr = (uint8_t*)data; uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len; size_t bufferLength = len;
// Dump the RTP packet to a file (if RTP dump is enabled).
if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP dump to output file failed");
}
int n = _transportPtr->SendPacket(channel, bufferToSendPtr, int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
bufferLength); bufferLength);
if (n < 0) { if (n < 0) {
@@ -290,14 +281,6 @@ Channel::SendRTCPPacket(int channel, const void *data, size_t len)
uint8_t* bufferToSendPtr = (uint8_t*)data; uint8_t* bufferToSendPtr = (uint8_t*)data;
size_t bufferLength = len; size_t bufferLength = len;
// Dump the RTCP packet to a file (if RTP dump is enabled).
if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTCP dump to output file failed");
}
int n = _transportPtr->SendRTCPPacket(channel, int n = _transportPtr->SendRTCPPacket(channel,
bufferToSendPtr, bufferToSendPtr,
bufferLength); bufferLength);
@@ -759,8 +742,6 @@ Channel::Channel(int32_t channelId,
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this, VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
this, this, rtp_payload_registry_.get())), this, this, rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
_outputAudioLevel(), _outputAudioLevel(),
_externalTransport(false), _externalTransport(false),
_inputFilePlayerPtr(NULL), _inputFilePlayerPtr(NULL),
@@ -927,8 +908,6 @@ Channel::~Channel()
// End of modules shutdown // End of modules shutdown
// Delete other objects // Delete other objects
RtpDump::DestroyRtpDump(&_rtpDumpIn);
RtpDump::DestroyRtpDump(&_rtpDumpOut);
delete &_callbackCritSect; delete &_callbackCritSect;
delete &_fileCritSect; delete &_fileCritSect;
delete &volume_settings_critsect_; delete &volume_settings_critsect_;
@@ -1640,13 +1619,6 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, size_t length,
// Store playout timestamp for the received RTP packet // Store playout timestamp for the received RTP packet
UpdatePlayoutTimestamp(false); UpdatePlayoutTimestamp(false);
// Dump the RTP packet to a file (if RTP dump is enabled).
if (_rtpDumpIn.DumpPacket((const uint8_t*)data,
(uint16_t)length) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP dump to input file failed");
}
const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data); const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data);
RTPHeader header; RTPHeader header;
if (!rtp_header_parser_->Parse(received_packet, length, &header)) { if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
@@ -1745,13 +1717,6 @@ int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
// Store playout timestamp for the received RTCP packet // Store playout timestamp for the received RTCP packet
UpdatePlayoutTimestamp(true); UpdatePlayoutTimestamp(true);
// Dump the RTCP packet to a file (if RTP dump is enabled).
if (_rtpDumpIn.DumpPacket((const uint8_t*)data, length) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice,
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTCP dump to input file failed");
}
// Deliver RTCP packet to RTP/RTCP module for parsing // Deliver RTCP packet to RTP/RTCP module for parsing
if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) { if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) {
_engineStatisticsPtr->SetLastError( _engineStatisticsPtr->SetLastError(
@@ -3372,82 +3337,6 @@ int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
return _rtpRtcpModule->SendNACK(sequence_numbers, length); return _rtpRtcpModule->SendNACK(sequence_numbers, length);
} }
int
Channel::StartRTPDump(const char fileNameUTF8[1024],
RTPDirections direction)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartRTPDump()");
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRTPDump() invalid RTP direction");
return -1;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
if (rtpDumpPtr == NULL)
{
assert(false);
return -1;
}
if (rtpDumpPtr->IsActive())
{
rtpDumpPtr->Stop();
}
if (rtpDumpPtr->Start(fileNameUTF8) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartRTPDump() failed to create file");
return -1;
}
return 0;
}
int
Channel::StopRTPDump(RTPDirections direction)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopRTPDump()");
if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StopRTPDump() invalid RTP direction");
return -1;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
if (rtpDumpPtr == NULL)
{
assert(false);
return -1;
}
if (!rtpDumpPtr->IsActive())
{
return 0;
}
return rtpDumpPtr->Stop();
}
bool
Channel::RTPDumpIsActive(RTPDirections direction)
{
if ((direction != kRtpIncoming) &&
(direction != kRtpOutgoing))
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"RTPDumpIsActive() invalid RTP direction");
return false;
}
RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
&_rtpDumpIn : &_rtpDumpOut;
return rtpDumpPtr->IsActive();
}
uint32_t uint32_t
Channel::Demultiplex(const AudioFrame& audioFrame) Channel::Demultiplex(const AudioFrame& audioFrame)
{ {

6
webrtc/voice_engine/channel.h
View File

@@ -51,7 +51,6 @@ class FileWrapper;
class ProcessThread; class ProcessThread;
class ReceiveStatistics; class ReceiveStatistics;
class RemoteNtpTimeEstimator; class RemoteNtpTimeEstimator;
class RtpDump;
class RTPPayloadRegistry; class RTPPayloadRegistry;
class RtpReceiver; class RtpReceiver;
class RTPReceiverAudio; class RTPReceiverAudio;
@@ -339,9 +338,6 @@ public:
int SetCodecFECStatus(bool enable); int SetCodecFECStatus(bool enable);
bool GetCodecFECStatus(); bool GetCodecFECStatus();
void SetNACKStatus(bool enable, int maxNumberOfPackets); void SetNACKStatus(bool enable, int maxNumberOfPackets);
int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
int StopRTPDump(RTPDirections direction);
bool RTPDumpIsActive(RTPDirections direction);
// From AudioPacketizationCallback in the ACM // From AudioPacketizationCallback in the ACM
int32_t SendData(FrameType frameType, int32_t SendData(FrameType frameType,
@@ -496,8 +492,6 @@ private:
TelephoneEventHandler* telephone_event_handler_; TelephoneEventHandler* telephone_event_handler_;
rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule; rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
rtc::scoped_ptr<AudioCodingModule> audio_coding_; rtc::scoped_ptr<AudioCodingModule> audio_coding_;
RtpDump& _rtpDumpIn;
RtpDump& _rtpDumpOut;
AudioLevel _outputAudioLevel; AudioLevel _outputAudioLevel;
bool _externalTransport; bool _externalTransport;
AudioFrame _audioFrame; AudioFrame _audioFrame;

18
webrtc/voice_engine/include/voe_rtp_rtcp.h
View File

@@ -221,24 +221,6 @@ class WEBRTC_DLLEXPORT VoERTP_RTCP {
// If using NACK, NACK should be enabled on both endpoints in a call. // If using NACK, NACK should be enabled on both endpoints in a call.
virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0; virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0;
// Enables capturing of RTP packets to a binary file on a specific
// |channel| and for a given |direction|. The file can later be replayed
// using e.g. RTP Tools rtpplay since the binary file format is
// compatible with the rtpdump format.
virtual int StartRTPDump(int channel,
const char fileNameUTF8[1024],
RTPDirections direction = kRtpIncoming) = 0;
// Disables capturing of RTP packets to a binary file on a specific
// |channel| and for a given |direction|.
virtual int StopRTPDump(int channel,
RTPDirections direction = kRtpIncoming) = 0;
// Gets the the current RTP capturing state for the specified
// |channel| and |direction|.
virtual int RTPDumpIsActive(int channel,
RTPDirections direction = kRtpIncoming) = 0;
// Will be removed. Don't use. // Will be removed. Don't use.
virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer) { virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer) {
return -1; return -1;

20
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
View File

@@ -117,23 +117,3 @@ TEST_F(RtpRtcpTest, DISABLED_ON_LINUX(SSRCPropagatesCorrectly)) {
EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc)); EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
EXPECT_EQ(local_ssrc, ssrc); EXPECT_EQ(local_ssrc, ssrc);
} }
// TODO(xians, phoglund): Re-enable when issue 372 is resolved.
TEST_F(RtpRtcpTest, DISABLED_CanCreateRtpDumpFilesWithoutError) {
// Create two RTP dump files (3 seconds long). You can verify these after
// the test using rtpplay or NetEqRTPplay if you like.
std::string output_path = webrtc::test::OutputPath();
std::string incoming_filename = output_path + "dump_in_3sec.rtp";
std::string outgoing_filename = output_path + "dump_out_3sec.rtp";
EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
channel_, incoming_filename.c_str(), webrtc::kRtpIncoming));
EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
channel_, outgoing_filename.c_str(), webrtc::kRtpOutgoing));
Sleep(3000);
EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpIncoming));
EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpOutgoing));
}

55
webrtc/voice_engine/voe_rtp_rtcp_impl.cc
View File

@@ -444,61 +444,6 @@ int VoERTP_RTCPImpl::SetNACKStatus(int channel, bool enable, int maxNoPackets) {
return 0; return 0;
} }
int VoERTP_RTCPImpl::StartRTPDump(int channel,
const char fileNameUTF8[1024],
RTPDirections direction) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"StartRTPDump(channel=%d, fileNameUTF8=%s, direction=%d)",
channel, fileNameUTF8, direction);
static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StartRTPDump() failed to locate channel");
return -1;
}
return channelPtr->StartRTPDump(fileNameUTF8, direction);
}
int VoERTP_RTCPImpl::StopRTPDump(int channel, RTPDirections direction) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"StopRTPDump(channel=%d, direction=%d)", channel, direction);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StopRTPDump() failed to locate channel");
return -1;
}
return channelPtr->StopRTPDump(direction);
}
int VoERTP_RTCPImpl::RTPDumpIsActive(int channel, RTPDirections direction) {
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"RTPDumpIsActive(channel=%d, direction=%d)", channel, direction);
if (!_shared->statistics().Initialized()) {
_shared->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == NULL) {
_shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"StopRTPDump() failed to locate channel");
return -1;
}
return channelPtr->RTPDumpIsActive(direction);
}
#endif // #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API #endif // #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
} // namespace webrtc } // namespace webrtc

10
webrtc/voice_engine/voe_rtp_rtcp_impl.h
View File

@@ -79,16 +79,6 @@ class VoERTP_RTCPImpl : public VoERTP_RTCP {
// NACK // NACK
int SetNACKStatus(int channel, bool enable, int maxNoPackets) override; int SetNACKStatus(int channel, bool enable, int maxNoPackets) override;
// Store RTP and RTCP packets and dump to file (compatible with rtpplay)
int StartRTPDump(int channel,
const char fileNameUTF8[1024],
RTPDirections direction = kRtpIncoming) override;
int StopRTPDump(int channel, RTPDirections direction = kRtpIncoming) override;
int RTPDumpIsActive(int channel,
RTPDirections direction = kRtpIncoming) override;
protected: protected:
VoERTP_RTCPImpl(voe::SharedData* shared); VoERTP_RTCPImpl(voe::SharedData* shared);
~VoERTP_RTCPImpl() override; ~VoERTP_RTCPImpl() override;