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Lint fix for webrtc/modules/video_coding PART 2!

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Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1543503002

Cr-Commit-Position: refs/heads/master@{#11102}
This commit is contained in:
philipel
2015-12-21 04:12:39 -08:00
committed by Commit bot
parent ff483617a4
commit 9d3ab61325
48 changed files with 3744 additions and 10210 deletions
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40
webrtc/modules/video_coding/receiver.cc
View File

@ -14,6 +14,7 @@
#include <cstdlib>
#include <utility>
#include <vector>
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
@ -72,8 +73,8 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
// Insert the packet into the jitter buffer. The packet can either be empty or
// contain media at this point.
bool retransmitted = false;
const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
&retransmitted);
const VCMFrameBufferEnum ret =
jitter_buffer_.InsertPacket(packet, &retransmitted);
if (ret == kOldPacket) {
return VCM_OK;
} else if (ret == kFlushIndicator) {
@ -96,13 +97,13 @@ void VCMReceiver::TriggerDecoderShutdown() {
}
VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
int64_t& next_render_time_ms,
int64_t* next_render_time_ms,
bool prefer_late_decoding) {
const int64_t start_time_ms = clock_->TimeInMilliseconds();
uint32_t frame_timestamp = 0;
// Exhaust wait time to get a complete frame for decoding.
bool found_frame = jitter_buffer_.NextCompleteTimestamp(
max_wait_time_ms, &frame_timestamp);
bool found_frame =
jitter_buffer_.NextCompleteTimestamp(max_wait_time_ms, &frame_timestamp);
if (!found_frame)
found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
@ -114,14 +115,14 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
const int64_t now_ms = clock_->TimeInMilliseconds();
timing_->UpdateCurrentDelay(frame_timestamp);
next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
*next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
// Check render timing.
bool timing_error = false;
// Assume that render timing errors are due to changes in the video stream.
if (next_render_time_ms < 0) {
if (*next_render_time_ms < 0) {
timing_error = true;
} else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms));
} else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
int frame_delay = static_cast<int>(std::abs(*next_render_time_ms - now_ms));
LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
<< "delay bounds (" << frame_delay << " > "
<< max_video_delay_ms_
@ -143,12 +144,13 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
if (prefer_late_decoding) {
// Decode frame as close as possible to the render timestamp.
const int32_t available_wait_time = max_wait_time_ms -
const int32_t available_wait_time =
max_wait_time_ms -
static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
uint16_t new_max_wait_time = static_cast<uint16_t>(
VCM_MAX(available_wait_time, 0));
uint16_t new_max_wait_time =
static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
uint32_t wait_time_ms = timing_->MaxWaitingTime(
next_render_time_ms, clock_->TimeInMilliseconds());
*next_render_time_ms, clock_->TimeInMilliseconds());
if (new_max_wait_time < wait_time_ms) {
// We're not allowed to wait until the frame is supposed to be rendered,
// waiting as long as we're allowed to avoid busy looping, and then return
@ -165,9 +167,9 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
if (frame == NULL) {
return NULL;
}
frame->SetRenderTime(next_render_time_ms);
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
"SetRenderTS", "render_time", next_render_time_ms);
frame->SetRenderTime(*next_render_time_ms);
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
"render_time", *next_render_time_ms);
if (!frame->Complete()) {
// Update stats for incomplete frames.
bool retransmitted = false;
@ -187,8 +189,7 @@ void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
jitter_buffer_.ReleaseFrame(frame);
}
void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
uint32_t* framerate) {
void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
assert(bitrate);
assert(framerate);
jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
@ -210,8 +211,7 @@ void VCMReceiver::SetNackMode(VCMNackMode nackMode,
void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
int max_packet_age_to_nack,
int max_incomplete_time_ms) {
jitter_buffer_.SetNackSettings(max_nack_list_size,
max_packet_age_to_nack,
jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
max_incomplete_time_ms);
}