webrtc_m130/webrtc/video/video_capture_input.cc
Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

170 lines
6.5 KiB
C++

/*
* 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/video/video_capture_input.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_capture/include/video_capture_factory.h"
#include "webrtc/modules/video_processing/main/interface/video_processing.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video/send_statistics_proxy.h"
#include "webrtc/video_engine/overuse_frame_detector.h"
#include "webrtc/video_engine/vie_encoder.h"
namespace webrtc {
namespace internal {
VideoCaptureInput::VideoCaptureInput(
ProcessThread* module_process_thread,
VideoCaptureCallback* frame_callback,
VideoRenderer* local_renderer,
SendStatisticsProxy* stats_proxy,
CpuOveruseObserver* overuse_observer,
EncodingTimeObserver* encoding_time_observer)
: capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
module_process_thread_(module_process_thread),
frame_callback_(frame_callback),
local_renderer_(local_renderer),
stats_proxy_(stats_proxy),
incoming_frame_cs_(CriticalSectionWrapper::CreateCriticalSection()),
encoder_thread_(ThreadWrapper::CreateThread(EncoderThreadFunction,
this,
"EncoderThread")),
capture_event_(EventWrapper::Create()),
stop_(0),
last_captured_timestamp_(0),
delta_ntp_internal_ms_(
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
TickTime::MillisecondTimestamp()),
overuse_detector_(new OveruseFrameDetector(Clock::GetRealTimeClock(),
CpuOveruseOptions(),
overuse_observer,
stats_proxy)),
encoding_time_observer_(encoding_time_observer) {
encoder_thread_->Start();
encoder_thread_->SetPriority(kHighPriority);
module_process_thread_->RegisterModule(overuse_detector_.get());
}
VideoCaptureInput::~VideoCaptureInput() {
module_process_thread_->DeRegisterModule(overuse_detector_.get());
// Stop the thread.
rtc::AtomicOps::ReleaseStore(&stop_, 1);
capture_event_->Set();
encoder_thread_->Stop();
}
void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
// TODO(pbos): Remove local rendering, it should be handled by the client code
// if required.
if (local_renderer_)
local_renderer_->RenderFrame(video_frame, 0);
stats_proxy_->OnIncomingFrame(video_frame.width(), video_frame.height());
VideoFrame incoming_frame = video_frame;
if (incoming_frame.ntp_time_ms() != 0) {
// If a NTP time stamp is set, this is the time stamp we will use.
incoming_frame.set_render_time_ms(incoming_frame.ntp_time_ms() -
delta_ntp_internal_ms_);
} else { // NTP time stamp not set.
int64_t render_time = incoming_frame.render_time_ms() != 0
? incoming_frame.render_time_ms()
: TickTime::MillisecondTimestamp();
incoming_frame.set_render_time_ms(render_time);
incoming_frame.set_ntp_time_ms(render_time + delta_ntp_internal_ms_);
}
// Convert NTP time, in ms, to RTP timestamp.
const int kMsToRtpTimestamp = 90;
incoming_frame.set_timestamp(
kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
CriticalSectionScoped cs(capture_cs_.get());
if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
// We don't allow the same capture time for two frames, drop this one.
LOG(LS_WARNING) << "Same/old NTP timestamp ("
<< incoming_frame.ntp_time_ms()
<< " <= " << last_captured_timestamp_
<< ") for incoming frame. Dropping.";
return;
}
captured_frame_.ShallowCopy(incoming_frame);
last_captured_timestamp_ = incoming_frame.ntp_time_ms();
overuse_detector_->FrameCaptured(captured_frame_.width(),
captured_frame_.height(),
captured_frame_.render_time_ms());
TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
"render_time", video_frame.render_time_ms());
capture_event_->Set();
}
bool VideoCaptureInput::EncoderThreadFunction(void* obj) {
return static_cast<VideoCaptureInput*>(obj)->EncoderProcess();
}
bool VideoCaptureInput::EncoderProcess() {
static const int kThreadWaitTimeMs = 100;
int64_t capture_time = -1;
if (capture_event_->Wait(kThreadWaitTimeMs) == kEventSignaled) {
if (rtc::AtomicOps::AcquireLoad(&stop_))
return false;
int64_t encode_start_time = -1;
VideoFrame deliver_frame;
{
CriticalSectionScoped cs(capture_cs_.get());
if (!captured_frame_.IsZeroSize()) {
deliver_frame = captured_frame_;
captured_frame_.Reset();
}
}
if (!deliver_frame.IsZeroSize()) {
capture_time = deliver_frame.render_time_ms();
encode_start_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
frame_callback_->DeliverFrame(deliver_frame);
}
// Update the overuse detector with the duration.
if (encode_start_time != -1) {
int encode_time_ms = static_cast<int>(
Clock::GetRealTimeClock()->TimeInMilliseconds() - encode_start_time);
overuse_detector_->FrameEncoded(encode_time_ms);
stats_proxy_->OnEncodedFrame(encode_time_ms);
if (encoding_time_observer_) {
encoding_time_observer_->OnReportEncodedTime(
deliver_frame.ntp_time_ms(), encode_time_ms);
}
}
}
// We're done!
if (capture_time != -1) {
overuse_detector_->FrameSent(capture_time);
}
return true;
}
} // namespace internal
} // namespace webrtc