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/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index 677f899..f729ddd 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -152,7 +152,7 @@
virtual int32 InitEncode(const webrtc::VideoCodec* codecSettings,
int32 numberOfCores,
- uint32 maxPayloadSize) {
+ size_t maxPayloadSize) {
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -351,7 +351,7 @@
// From ViEExternalCapture
virtual int IncomingFrame(unsigned char* videoFrame,
- unsigned int videoFrameLength,
+ size_t videoFrameLength,
unsigned short width,
unsigned short height,
webrtc::RawVideoType videoType,
@@ -890,7 +890,7 @@
WEBRTC_FUNC(ReceivedRTPPacket, (const int channel,
const void* packet,
- const int length,
+ const size_t length,
const webrtc::PacketTime& packet_time)) {
WEBRTC_ASSERT_CHANNEL(channel);
ASSERT(length > 1);
@@ -899,11 +899,11 @@
return 0;
}
- WEBRTC_STUB(ReceivedRTCPPacket, (const int, const void*, const int));
+ WEBRTC_STUB(ReceivedRTCPPacket, (const int, const void*, const size_t));
// Not using WEBRTC_STUB due to bool return value
virtual bool IsIPv6Enabled(int channel) { return true; }
WEBRTC_STUB(SetMTU, (int, unsigned int));
- WEBRTC_STUB(ReceivedBWEPacket, (const int, int64_t, int,
+ WEBRTC_STUB(ReceivedBWEPacket, (const int, int64_t, size_t,
const webrtc::RTPHeader&));
virtual bool SetBandwidthEstimationConfig(int, const webrtc::Config&) {
return true;
@@ -1140,8 +1140,8 @@
unsigned int&, unsigned int&, unsigned int&, int&));
WEBRTC_STUB_CONST(GetSentRTCPStatistics, (const int, unsigned short&,
unsigned int&, unsigned int&, unsigned int&, int&));
- WEBRTC_STUB_CONST(GetRTPStatistics, (const int, unsigned int&, unsigned int&,
- unsigned int&, unsigned int&));
+ WEBRTC_STUB_CONST(GetRTPStatistics, (const int, size_t&, unsigned int&,
+ size_t&, unsigned int&));
WEBRTC_STUB_CONST(GetReceiveChannelRtcpStatistics, (const int,
webrtc::RtcpStatistics&, int&));
WEBRTC_STUB_CONST(GetSendChannelRtcpStatistics, (const int,
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 222ce4b..dc27e96 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -858,7 +858,7 @@
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- unsigned int length)) {
+ size_t length)) {
WEBRTC_CHECK_CHANNEL(channel);
if (!channels_[channel]->external_transport) return -1;
channels_[channel]->packets.push_back(
@@ -866,7 +866,7 @@
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- unsigned int length,
+ size_t length,
const webrtc::PacketTime& packet_time)) {
WEBRTC_CHECK_CHANNEL(channel);
if (ReceivedRTPPacket(channel, data, length) == -1) {
@@ -877,7 +877,7 @@
}
WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
- unsigned int length));
+ size_t length));
// webrtc::VoERTP_RTCP
WEBRTC_STUB(RegisterRTPObserver, (int channel,
diff --git a/talk/media/webrtc/webrtcvideocapturer.cc b/talk/media/webrtc/webrtcvideocapturer.cc
index ea8b61a..32f105d 100644
--- a/talk/media/webrtc/webrtcvideocapturer.cc
+++ b/talk/media/webrtc/webrtcvideocapturer.cc
@@ -36,6 +36,7 @@
#include "talk/media/webrtc/webrtcvideoframefactory.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
@@ -351,8 +352,8 @@
// Signal down stream components on captured frame.
// The CapturedFrame class doesn't support planes. We have to ExtractBuffer
// to one block for it.
- int length = webrtc::CalcBufferSize(webrtc::kI420,
- sample.width(), sample.height());
+ size_t length =
+ webrtc::CalcBufferSize(webrtc::kI420, sample.width(), sample.height());
capture_buffer_.resize(length);
// TODO(ronghuawu): Refactor the WebRtcCapturedFrame to avoid memory copy.
webrtc::ExtractBuffer(sample, length, &capture_buffer_[0]);
@@ -368,7 +369,7 @@
// WebRtcCapturedFrame
WebRtcCapturedFrame::WebRtcCapturedFrame(const webrtc::I420VideoFrame& sample,
void* buffer,
- int length) {
+ size_t length) {
width = sample.width();
height = sample.height();
fourcc = FOURCC_I420;
@@ -378,7 +379,7 @@
// Convert units from VideoFrame RenderTimeMs to CapturedFrame (nanoseconds).
elapsed_time = sample.render_time_ms() * rtc::kNumNanosecsPerMillisec;
time_stamp = elapsed_time;
- data_size = length;
+ data_size = rtc::checked_cast<uint32>(length);
data = buffer;
}
diff --git a/talk/media/webrtc/webrtcvideocapturer.h b/talk/media/webrtc/webrtcvideocapturer.h
index 39a71f8..7d09040 100644
--- a/talk/media/webrtc/webrtcvideocapturer.h
+++ b/talk/media/webrtc/webrtcvideocapturer.h
@@ -98,7 +98,7 @@
struct WebRtcCapturedFrame : public CapturedFrame {
public:
WebRtcCapturedFrame(const webrtc::I420VideoFrame& frame,
- void* buffer, int length);
+ void* buffer, size_t length);
};
} // namespace cricket
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 88acc3f..1e8c43b 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -305,7 +305,7 @@
}
virtual int DeliverFrame(unsigned char* buffer,
- int buffer_size,
+ size_t buffer_size,
uint32_t rtp_time_stamp,
int64_t ntp_time_ms,
int64_t render_time,
@@ -347,14 +347,14 @@
virtual bool IsTextureSupported() { return true; }
- int DeliverBufferFrame(unsigned char* buffer, int buffer_size,
+ int DeliverBufferFrame(unsigned char* buffer, size_t buffer_size,
int64 time_stamp, int64 elapsed_time) {
WebRtcVideoFrame video_frame;
video_frame.Alias(buffer, buffer_size, width_, height_,
1, 1, elapsed_time, time_stamp, 0);
// Sanity check on decoded frame size.
- if (buffer_size != static_cast<int>(VideoFrame::SizeOf(width_, height_))) {
+ if (buffer_size != VideoFrame::SizeOf(width_, height_)) {
LOG(LS_WARNING) << "WebRtcRenderAdapter (channel " << channel_id_
<< ") received a strange frame size: "
<< buffer_size;
@@ -2499,7 +2499,8 @@
ASSERT(channel_id == default_channel_id_);
continue;
}
- unsigned int bytes_sent, packets_sent, bytes_recv, packets_recv;
+ size_t bytes_sent, bytes_recv;
+ unsigned int packets_sent, packets_recv;
if (engine_->vie()->rtp()->GetRTPStatistics(channel_id, bytes_sent,
packets_sent, bytes_recv,
packets_recv) != 0) {
@@ -2829,7 +2830,7 @@
engine()->vie()->network()->ReceivedRTPPacket(
processing_channel_id,
packet->data(),
- static_cast<int>(packet->length()),
+ packet->length(),
webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
@@ -2858,7 +2859,7 @@
engine_->vie()->network()->ReceivedRTCPPacket(
recv_channel_id,
packet->data(),
- static_cast<int>(packet->length()));
+ packet->length());
}
}
// SR may continue RR and any RR entry may correspond to any one of the send
@@ -2871,7 +2872,7 @@
engine_->vie()->network()->ReceivedRTCPPacket(
channel_id,
packet->data(),
- static_cast<int>(packet->length()));
+ packet->length());
}
}
@@ -4022,16 +4023,16 @@
}
int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data,
- int len) {
+ size_t len) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
- return MediaChannel::SendPacket(&packet) ? len : -1;
+ return MediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
}
int WebRtcVideoMediaChannel::SendRTCPPacket(int channel,
const void* data,
- int len) {
+ size_t len) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
- return MediaChannel::SendRtcp(&packet) ? len : -1;
+ return MediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
}
void WebRtcVideoMediaChannel::QueueBlackFrame(uint32 ssrc, int64 timestamp,
diff --git a/talk/media/webrtc/webrtcvideoengine.h b/talk/media/webrtc/webrtcvideoengine.h
index db091af..d1ace7d 100644
--- a/talk/media/webrtc/webrtcvideoengine.h
+++ b/talk/media/webrtc/webrtcvideoengine.h
@@ -331,8 +331,10 @@
int GetLastEngineError() { return engine()->GetLastEngineError(); }
// webrtc::Transport:
- virtual int SendPacket(int channel, const void* data, int len) OVERRIDE;
- virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE;
+ virtual int SendPacket(int channel, const void* data, size_t len) OVERRIDE;
+ virtual int SendRTCPPacket(int channel,
+ const void* data,
+ size_t len) OVERRIDE;
bool ConferenceModeIsEnabled() const {
return options_.conference_mode.GetWithDefaultIfUnset(false);
diff --git a/talk/media/webrtc/webrtcvideoframe.cc b/talk/media/webrtc/webrtcvideoframe.cc
index 9e4ea06..9dbf5a5 100644
--- a/talk/media/webrtc/webrtcvideoframe.cc
+++ b/talk/media/webrtc/webrtcvideoframe.cc
@@ -71,8 +71,8 @@
// Make sure that |video_frame_| doesn't delete the buffer, as |owned_data_|
// will release the buffer if this FrameBuffer owns it.
uint8_t* new_memory = NULL;
- uint32_t new_length = 0;
- uint32_t new_size = 0;
+ size_t new_length = 0;
+ size_t new_size = 0;
video_frame_.Swap(new_memory, new_length, new_size);
}
@@ -84,8 +84,8 @@
void WebRtcVideoFrame::FrameBuffer::Alias(uint8* data, size_t length) {
owned_data_.reset();
uint8_t* new_memory = reinterpret_cast<uint8_t*>(data);
- uint32_t new_length = static_cast<uint32_t>(length);
- uint32_t new_size = static_cast<uint32_t>(length);
+ size_t new_length = length;
+ size_t new_size = length;
video_frame_.Swap(new_memory, new_length, new_size);
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index c335626..9f5d306 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -3154,7 +3154,7 @@
engine()->voe()->network()->ReceivedRTPPacket(
which_channel,
packet->data(),
- static_cast<unsigned int>(packet->length()),
+ packet->length(),
webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
@@ -3179,7 +3179,7 @@
engine()->voe()->network()->ReceivedRTCPPacket(
which_channel,
packet->data(),
- static_cast<unsigned int>(packet->length()));
+ packet->length());
if (IsDefaultChannel(which_channel))
has_sent_to_default_channel = true;
@@ -3199,7 +3199,7 @@
engine()->voe()->network()->ReceivedRTCPPacket(
iter->second->channel(),
packet->data(),
- static_cast<unsigned int>(packet->length()));
+ packet->length());
}
}
@@ -3730,7 +3730,7 @@
return true;
}
-int WebRtcSoundclipStream::Read(void *buf, int len) {
+int WebRtcSoundclipStream::Read(void *buf, size_t len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
return static_cast<int>(res);
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 67fadc5..6dc1f25 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -69,7 +69,7 @@
}
void set_loop(bool loop) { loop_ = loop; }
- virtual int Read(void* buf, int len) OVERRIDE;
+ virtual int Read(void* buf, size_t len) OVERRIDE;
virtual int Rewind() OVERRIDE;
private:
@@ -80,7 +80,7 @@
// WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
// For now we just dump the data.
class WebRtcMonitorStream : public webrtc::OutStream {
- virtual bool Write(const void *buf, int len) OVERRIDE {
+ virtual bool Write(const void *buf, size_t len) OVERRIDE {
return true;
}
};
@@ -315,17 +315,16 @@
protected:
// implements Transport interface
- virtual int SendPacket(int channel, const void *data, int len) OVERRIDE {
+ virtual int SendPacket(int channel, const void *data, size_t len) OVERRIDE {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
- if (!T::SendPacket(&packet)) {
- return -1;
- }
- return len;
+ return T::SendPacket(&packet) ? static_cast<int>(len) : -1;
}
- virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE {
+ virtual int SendRTCPPacket(int channel,
+ const void* data,
+ size_t len) OVERRIDE {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
- return T::SendRtcp(&packet) ? len : -1;
+ return T::SendRtcp(&packet) ? static_cast<int>(len) : -1;
}
private: