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Use size_t more consistently for packet/payload lengths.

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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
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
talk/app/webrtc/java/jni/peerconnection_jni.cc
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@@ -1258,7 +1258,7 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder,
// |codec_thread_| for execution.
virtual int32_t InitEncode(const webrtc::VideoCodec* codec_settings,
int32_t /* number_of_cores */,
uint32_t /* max_payload_size */) OVERRIDE;
size_t /* max_payload_size */) OVERRIDE;
virtual int32_t Encode(
const webrtc::I420VideoFrame& input_image,
const webrtc::CodecSpecificInfo* /* codec_specific_info */,
@@ -1433,7 +1433,7 @@ MediaCodecVideoEncoder::MediaCodecVideoEncoder(JNIEnv* jni)
int32_t MediaCodecVideoEncoder::InitEncode(
const webrtc::VideoCodec* codec_settings,
int32_t /* number_of_cores */,
uint32_t /* max_payload_size */) {
size_t /* max_payload_size */) {
// Factory should guard against other codecs being used with us.
CHECK(codec_settings->codecType == kVideoCodecVP8) << "Unsupported codec";

14
talk/media/webrtc/fakewebrtcvideoengine.h
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@@ -152,7 +152,7 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
virtual int32 InitEncode(const webrtc::VideoCodec* codecSettings,
int32 numberOfCores,
uint32 maxPayloadSize) {
size_t maxPayloadSize) {
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -351,7 +351,7 @@ class FakeWebRtcVideoEngine
// 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 @@ class FakeWebRtcVideoEngine
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 @@ class FakeWebRtcVideoEngine
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 @@ class FakeWebRtcVideoEngine
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,

6
talk/media/webrtc/fakewebrtcvoiceengine.h
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@@ -858,7 +858,7 @@ class FakeWebRtcVoiceEngine
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 @@ class FakeWebRtcVoiceEngine
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 @@ class FakeWebRtcVoiceEngine
}
WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
unsigned int length));
size_t length));
// webrtc::VoERTP_RTCP
WEBRTC_STUB(RegisterRTPObserver, (int channel,

9
talk/media/webrtc/webrtcvideocapturer.cc
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@@ -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 @@ void WebRtcVideoCapturer::OnIncomingCapturedFrame(const int32_t id,
// 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 @@ void WebRtcVideoCapturer::OnCaptureDelayChanged(const int32_t id,
// 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 @@ WebRtcCapturedFrame::WebRtcCapturedFrame(const webrtc::I420VideoFrame& sample,
// 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;
}

2
talk/media/webrtc/webrtcvideocapturer.h
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@@ -98,7 +98,7 @@ class WebRtcVideoCapturer : public VideoCapturer,
struct WebRtcCapturedFrame : public CapturedFrame {
public:
WebRtcCapturedFrame(const webrtc::I420VideoFrame& frame,
void* buffer, int length);
void* buffer, size_t length);
};
} // namespace cricket

23
talk/media/webrtc/webrtcvideoengine.cc
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@@ -305,7 +305,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
}
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 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
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 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options,
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 @@ void WebRtcVideoMediaChannel::OnPacketReceived(
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 @@ void WebRtcVideoMediaChannel::OnRtcpReceived(
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 @@ void WebRtcVideoMediaChannel::OnRtcpReceived(
engine_->vie()->network()->ReceivedRTCPPacket(
channel_id,
packet->data(),
static_cast<int>(packet->length()));
packet->length());
}
}
@@ -4022,16 +4023,16 @@ void WebRtcVideoMediaChannel::OnMessage(rtc::Message* msg) {
}
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,

6
talk/media/webrtc/webrtcvideoengine.h
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@@ -331,8 +331,10 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
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);

8
talk/media/webrtc/webrtcvideoframe.cc
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@@ -71,8 +71,8 @@ WebRtcVideoFrame::FrameBuffer::~FrameBuffer() {
// 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::Attach(uint8* data, size_t length) {
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);
}

8
talk/media/webrtc/webrtcvoiceengine.cc
View File

@@ -3154,7 +3154,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
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 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
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 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
engine()->voe()->network()->ReceivedRTCPPacket(
iter->second->channel(),
packet->data(),
static_cast<unsigned int>(packet->length()));
packet->length());
}
}
@@ -3730,7 +3730,7 @@ bool WebRtcVoiceMediaChannel::SetupSharedBweOnChannel(int voe_channel) {
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);

17
talk/media/webrtc/webrtcvoiceengine.h
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@@ -69,7 +69,7 @@ class WebRtcSoundclipStream : public webrtc::InStream {
}
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 @@ class WebRtcSoundclipStream : public webrtc::InStream {
// 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 @@ class WebRtcMediaChannel : public T, public webrtc::Transport {
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: