mirror of
https://github.com/classilla/tenfourfox.git
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190 lines
6.2 KiB
C++
190 lines
6.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/vie_sync_module.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/trace_event.h"
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#include "webrtc/video_engine/stream_synchronization.h"
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#include "webrtc/video_engine/vie_channel.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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int UpdateMeasurements(StreamSynchronization::Measurements* stream,
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const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
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if (!receiver.Timestamp(&stream->latest_timestamp))
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return -1;
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if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
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return -1;
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uint32_t ntp_secs = 0;
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uint32_t ntp_frac = 0;
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uint32_t rtp_timestamp = 0;
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if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
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&ntp_frac,
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NULL,
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NULL,
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&rtp_timestamp)) {
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return -1;
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}
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bool new_rtcp_sr = false;
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if (!UpdateRtcpList(
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ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
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return -1;
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}
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return 0;
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}
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ViESyncModule::ViESyncModule(VideoCodingModule* vcm,
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ViEChannel* vie_channel)
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: data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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vcm_(vcm),
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vie_channel_(vie_channel),
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video_receiver_(NULL),
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video_rtp_rtcp_(NULL),
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voe_channel_id_(-1),
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voe_sync_interface_(NULL),
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last_sync_time_(TickTime::Now()),
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sync_() {
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}
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ViESyncModule::~ViESyncModule() {
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}
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int ViESyncModule::ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface,
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RtpRtcp* video_rtcp_module,
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RtpReceiver* video_receiver) {
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CriticalSectionScoped cs(data_cs_.get());
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voe_channel_id_ = voe_channel_id;
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voe_sync_interface_ = voe_sync_interface;
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video_receiver_ = video_receiver;
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video_rtp_rtcp_ = video_rtcp_module;
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sync_.reset(new StreamSynchronization(voe_channel_id, vie_channel_->Id()));
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if (!voe_sync_interface) {
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voe_channel_id_ = -1;
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if (voe_channel_id >= 0) {
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// Trying to set a voice channel but no interface exist.
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return -1;
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}
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return 0;
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}
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return 0;
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}
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int ViESyncModule::VoiceChannel() {
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return voe_channel_id_;
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}
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int64_t ViESyncModule::TimeUntilNextProcess() {
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const int64_t kSyncIntervalMs = 1000;
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return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
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}
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int32_t ViESyncModule::Process() {
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CriticalSectionScoped cs(data_cs_.get());
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last_sync_time_ = TickTime::Now();
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const int current_video_delay_ms = vcm_->Delay();
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if (voe_channel_id_ == -1) {
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return 0;
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}
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assert(video_rtp_rtcp_ && voe_sync_interface_);
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assert(sync_.get());
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int audio_jitter_buffer_delay_ms = 0;
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int playout_buffer_delay_ms = 0;
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int avsync_offset_ms = 0;
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if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
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&audio_jitter_buffer_delay_ms,
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&playout_buffer_delay_ms,
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&avsync_offset_ms) != 0) {
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return 0;
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}
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const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
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playout_buffer_delay_ms;
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RtpRtcp* voice_rtp_rtcp = NULL;
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RtpReceiver* voice_receiver = NULL;
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if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
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&voice_receiver)) {
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return 0;
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}
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assert(voice_rtp_rtcp);
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assert(voice_receiver);
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if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
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*video_receiver_) != 0) {
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return 0;
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}
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if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
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*voice_receiver) != 0) {
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return 0;
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}
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int relative_delay_ms;
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int result;
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// Calculate how much later or earlier the audio stream is compared to video.
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result = sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
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&relative_delay_ms);
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if (!result) {
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return 0;
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}
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voe_sync_interface_->SetCurrentSyncOffset(voe_channel_id_, relative_delay_ms);
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TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
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TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
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TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
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int target_audio_delay_ms = 0;
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int target_video_delay_ms = current_video_delay_ms;
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// Calculate the necessary extra audio delay and desired total video
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// delay to get the streams in sync.
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if (!sync_->ComputeDelays(relative_delay_ms,
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current_audio_delay_ms,
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&target_audio_delay_ms,
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&target_video_delay_ms)) {
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return 0;
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}
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if (voe_sync_interface_->SetMinimumPlayoutDelay(
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voe_channel_id_, target_audio_delay_ms) == -1) {
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LOG(LS_ERROR) << "Error setting voice delay.";
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}
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vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
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return 0;
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}
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int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
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CriticalSectionScoped cs(data_cs_.get());
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if (!voe_sync_interface_) {
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LOG(LS_ERROR) << "voe_sync_interface_ NULL, can't set playout delay.";
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return -1;
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}
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sync_->SetTargetBufferingDelay(target_delay_ms);
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// Setting initial playout delay to voice engine (video engine is updated via
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// the VCM interface).
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voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
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target_delay_ms);
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return 0;
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}
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} // namespace webrtc
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