Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/utility/interface/rtp_dump.h b/webrtc/modules/utility/interface/rtp_dump.h
index 6c2dc7c..df45ae2 100644
--- a/webrtc/modules/utility/interface/rtp_dump.h
+++ b/webrtc/modules/utility/interface/rtp_dump.h
@@ -43,7 +43,7 @@
// 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,
- uint16_t packetLength) = 0;
+ size_t packetLength) = 0;
protected:
virtual ~RtpDump();
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 11a462a..5a39748 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -54,7 +54,7 @@
int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
uint32_t sampFreqHz,
const int8_t* incomingPayload,
- int32_t payloadLength)
+ size_t payloadLength)
{
if (payloadLength > 0)
{
@@ -79,7 +79,7 @@
int32_t AudioCoder::Encode(const AudioFrame& audio,
int8_t* encodedData,
- uint32_t& encodedLengthInBytes)
+ size_t& encodedLengthInBytes)
{
// Fake a timestamp in case audio doesn't contain a correct timestamp.
// Make a local copy of the audio frame since audio is const
@@ -109,7 +109,7 @@
uint8_t /* payloadType */,
uint32_t /* timeStamp */,
const uint8_t* payloadData,
- uint16_t payloadSize,
+ size_t payloadSize,
const RTPFragmentationHeader* /* fragmentation*/)
{
memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
index e04372d..0363690 100644
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/modules/utility/source/coder.h
@@ -34,12 +34,12 @@
ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
- const int8_t* incomingPayload, int32_t payloadLength);
+ const int8_t* incomingPayload, size_t payloadLength);
int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
- uint32_t& encodedLengthInBytes);
+ size_t& encodedLengthInBytes);
protected:
virtual int32_t SendData(
@@ -47,7 +47,7 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- uint16_t payloadSize,
+ size_t payloadSize,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
@@ -57,7 +57,7 @@
uint32_t _encodeTimestamp;
int8_t* _encodedData;
- uint32_t _encodedLengthInBytes;
+ size_t _encodedLengthInBytes;
uint32_t _decodeTimestamp;
};
diff --git a/webrtc/modules/utility/source/file_player_impl.cc b/webrtc/modules/utility/source/file_player_impl.cc
index 8049245..29f7042 100644
--- a/webrtc/modules/utility/source/file_player_impl.cc
+++ b/webrtc/modules/utility/source/file_player_impl.cc
@@ -124,7 +124,7 @@
unresampledAudioFrame.sample_rate_hz_ = _codec.plfreq;
// L16 is un-encoded data. Just pull 10 ms.
- uint32_t lengthInBytes =
+ size_t lengthInBytes =
sizeof(unresampledAudioFrame.data_);
if (_fileModule.PlayoutAudioData(
(int8_t*)unresampledAudioFrame.data_,
@@ -147,11 +147,11 @@
// expects a full frame. If the frame size is larger than 10 ms,
// PlayoutAudioData(..) data should be called proportionally less often.
int16_t encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
- uint32_t encodedLengthInBytes = 0;
+ size_t encodedLengthInBytes = 0;
if(++_numberOf10MsInDecoder >= _numberOf10MsPerFrame)
{
_numberOf10MsInDecoder = 0;
- uint32_t bytesFromFile = sizeof(encodedBuffer);
+ size_t bytesFromFile = sizeof(encodedBuffer);
if (_fileModule.PlayoutAudioData((int8_t*)encodedBuffer,
bytesFromFile) == -1)
{
@@ -581,7 +581,7 @@
if(_fileFormat == kFileFormatAviFile)
{
// Get next video frame
- uint32_t encodedBufferLengthInBytes = _encodedData.bufferSize;
+ size_t encodedBufferLengthInBytes = _encodedData.bufferSize;
if(_fileModule.PlayoutAVIVideoData(
reinterpret_cast< int8_t*>(_encodedData.payloadData),
encodedBufferLengthInBytes) != 0)
@@ -656,7 +656,7 @@
// Size of unencoded data (I420) should be the largest possible frame size
// in a file.
- const uint32_t KReadBufferSize = 3 * video_codec_info_.width *
+ const size_t KReadBufferSize = 3 * video_codec_info_.width *
video_codec_info_.height / 2;
_encodedData.VerifyAndAllocate(KReadBufferSize);
_encodedData.encodedHeight = video_codec_info_.height;
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index 264b867..14c5288 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -227,7 +227,7 @@
// NOTE: stereo recording is only supported for WAV files.
// TODO (hellner): WAV expect PCM in little endian byte order. Not
// "encoding" with PCM coder should be a problem for big endian systems.
- uint32_t encodedLenInBytes = 0;
+ size_t encodedLenInBytes = 0;
if (_fileFormat == kFileFormatPreencodedFile ||
STR_CASE_CMP(codec_info_.plname, "L16") != 0)
{
@@ -272,9 +272,8 @@
uint16_t msOfData =
ptrAudioFrame->samples_per_channel_ /
uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000);
- if (WriteEncodedAudioData(_audioBuffer,
- (uint16_t)encodedLenInBytes,
- msOfData, playoutTS) == -1)
+ if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes, msOfData,
+ playoutTS) == -1)
{
return -1;
}
@@ -309,7 +308,7 @@
int32_t FileRecorderImpl::WriteEncodedAudioData(
const int8_t* audioBuffer,
- uint16_t bufferLength,
+ size_t bufferLength,
uint16_t /*millisecondsOfData*/,
const TickTime* /*playoutTS*/)
{
@@ -398,7 +397,7 @@
return FileRecorderImpl::StopRecording();
}
-int32_t AviRecorder::CalcI420FrameSize( ) const
+size_t AviRecorder::CalcI420FrameSize( ) const
{
return 3 * _videoCodecInst.width * _videoCodecInst.height / 2;
}
@@ -641,8 +640,8 @@
if( STR_CASE_CMP(_videoCodecInst.plName, "I420") == 0)
{
- int length = CalcBufferSize(kI420, videoFrame.width(),
- videoFrame.height());
+ size_t length =
+ CalcBufferSize(kI420, videoFrame.width(), videoFrame.height());
_videoEncodedData.VerifyAndAllocate(length);
// I420 is raw data. No encoding needed (each sample is represented by
@@ -681,7 +680,7 @@
// happens in AviRecorder::Process().
int32_t AviRecorder::WriteEncodedAudioData(
const int8_t* audioBuffer,
- uint16_t bufferLength,
+ size_t bufferLength,
uint16_t millisecondsOfData,
const TickTime* playoutTS)
{
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
index 53fd26b..9e17fd6 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/webrtc/modules/utility/source/file_recorder_impl.h
@@ -86,7 +86,7 @@
protected:
virtual int32_t WriteEncodedAudioData(
const int8_t* audioBuffer,
- uint16_t bufferLength,
+ size_t bufferLength,
uint16_t millisecondsOfData,
const TickTime* playoutTS);
@@ -111,7 +111,7 @@
{
public:
AudioFrameFileInfo(const int8_t* audioData,
- const uint16_t audioSize,
+ const size_t audioSize,
const uint16_t audioMS,
const TickTime& playoutTS)
: _audioData(), _audioSize(audioSize), _audioMS(audioMS),
@@ -127,7 +127,7 @@
};
// TODO (hellner): either turn into a struct or provide get/set functions.
int8_t _audioData[MAX_AUDIO_BUFFER_IN_BYTES];
- uint16_t _audioSize;
+ size_t _audioSize;
uint16_t _audioMS;
TickTime _playoutTS;
};
@@ -151,7 +151,7 @@
protected:
virtual int32_t WriteEncodedAudioData(
const int8_t* audioBuffer,
- uint16_t bufferLength,
+ size_t bufferLength,
uint16_t millisecondsOfData,
const TickTime* playoutTS);
private:
@@ -165,7 +165,7 @@
int32_t EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame);
int32_t ProcessAudio();
- int32_t CalcI420FrameSize() const;
+ size_t CalcI420FrameSize() const;
int32_t SetUpVideoEncoder();
VideoCodec _videoCodecInst;
@@ -178,7 +178,7 @@
FrameScaler* _frameScaler;
VideoCoder* _videoEncoder;
- int32_t _videoMaxPayloadSize;
+ size_t _videoMaxPayloadSize;
EncodedVideoData _videoEncodedData;
ThreadWrapper* _thread;
diff --git a/webrtc/modules/utility/source/rtp_dump_impl.cc b/webrtc/modules/utility/source/rtp_dump_impl.cc
index 547df33..cab4065 100644
--- a/webrtc/modules/utility/source/rtp_dump_impl.cc
+++ b/webrtc/modules/utility/source/rtp_dump_impl.cc
@@ -12,6 +12,7 @@
#include <assert.h>
#include <stdio.h>
+#include <limits>
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
@@ -145,7 +146,7 @@
return _file.Open();
}
-int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength)
+int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, size_t packetLength)
{
CriticalSectionScoped lock(_critSect);
if (!IsActive())
@@ -158,7 +159,9 @@
return -1;
}
- if (packetLength < 1)
+ rtpDumpPktHdr_t hdr;
+ size_t total_size = packetLength + sizeof hdr;
+ if (packetLength < 1 || total_size > std::numeric_limits<uint16_t>::max())
{
return -1;
}
@@ -167,11 +170,8 @@
// considered RTP (without further verification).
bool isRTCP = RTCP(packet);
- rtpDumpPktHdr_t hdr;
- uint32_t offset;
-
// Offset is relative to when recording was started.
- offset = GetTimeInMS();
+ uint32_t offset = GetTimeInMS();
if (offset < _startTime)
{
// Compensate for wraparound.
@@ -181,7 +181,7 @@
}
hdr.offset = RtpDumpHtonl(offset);
- hdr.length = RtpDumpHtons((uint16_t)(packetLength + sizeof(hdr)));
+ hdr.length = RtpDumpHtons((uint16_t)(total_size));
if (isRTCP)
{
hdr.plen = 0;
diff --git a/webrtc/modules/utility/source/rtp_dump_impl.h b/webrtc/modules/utility/source/rtp_dump_impl.h
index 0b72132..5be9cbe 100644
--- a/webrtc/modules/utility/source/rtp_dump_impl.h
+++ b/webrtc/modules/utility/source/rtp_dump_impl.h
@@ -26,7 +26,7 @@
virtual int32_t Stop() OVERRIDE;
virtual bool IsActive() const OVERRIDE;
virtual int32_t DumpPacket(const uint8_t* packet,
- uint16_t packetLength) OVERRIDE;
+ size_t packetLength) OVERRIDE;
private:
// Return the system time in ms.
inline uint32_t GetTimeInMS() const;
diff --git a/webrtc/modules/utility/source/video_coder.cc b/webrtc/modules/utility/source/video_coder.cc
index 5096ace..e0d969d 100644
--- a/webrtc/modules/utility/source/video_coder.cc
+++ b/webrtc/modules/utility/source/video_coder.cc
@@ -113,7 +113,7 @@
const uint32_t timeStamp,
int64_t capture_time_ms,
const uint8_t* payloadData,
- uint32_t payloadSize,
+ size_t payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
const RTPVideoHeader* /*rtpVideoHdr*/)
{
diff --git a/webrtc/modules/utility/source/video_coder.h b/webrtc/modules/utility/source/video_coder.h
index 03aa511..a1d1a17 100644
--- a/webrtc/modules/utility/source/video_coder.h
+++ b/webrtc/modules/utility/source/video_coder.h
@@ -53,7 +53,7 @@
uint32_t /*timeStamp*/,
int64_t capture_time_ms,
const uint8_t* payloadData,
- uint32_t payloadSize,
+ size_t payloadSize,
const RTPFragmentationHeader& /* fragmentationHeader*/,
const RTPVideoHeader* rtpTypeHdr) OVERRIDE;