sprang@webrtc.org dcebf2daa7 Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00

1007 lines
33 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_engine/vie_encoder.h"
#include <assert.h>
#include <algorithm>
#include "webrtc/common_video/interface/video_image.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/process_thread.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video_engine/include/vie_codec.h"
#include "webrtc/video_engine/include/vie_image_process.h"
#include "webrtc/frame_callback.h"
#include "webrtc/video_engine/vie_defines.h"
namespace webrtc {
// Margin on when we pause the encoder when the pacing buffer overflows relative
// to the configured buffer delay.
static const float kEncoderPausePacerMargin = 2.0f;
// Don't stop the encoder unless the delay is above this configured value.
static const int kMinPacingDelayMs = 200;
// Allow packets to be transmitted in up to 2 times max video bitrate if the
// bandwidth estimate allows it.
// TODO(holmer): Expose transmission start, min and max bitrates in the
// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
static const int kTransmissionMaxBitrateMultiplier = 2;
static const float kStopPaddingThresholdMs = 2000;
std::vector<uint32_t> AllocateStreamBitrates(
uint32_t total_bitrate,
const SimulcastStream* stream_configs,
size_t number_of_streams) {
if (number_of_streams == 0) {
std::vector<uint32_t> stream_bitrates(1, 0);
stream_bitrates[0] = total_bitrate;
return stream_bitrates;
}
std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
uint32_t bitrate_remainder = total_bitrate;
for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
stream_bitrates[i] = bitrate_remainder;
} else {
stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
}
bitrate_remainder -= stream_bitrates[i];
}
return stream_bitrates;
}
class QMVideoSettingsCallback : public VCMQMSettingsCallback {
public:
explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
~QMVideoSettingsCallback();
// Update VPM with QM (quality modes: frame size & frame rate) settings.
int32_t SetVideoQMSettings(const uint32_t frame_rate,
const uint32_t width,
const uint32_t height);
private:
VideoProcessingModule* vpm_;
};
class ViEBitrateObserver : public BitrateObserver {
public:
explicit ViEBitrateObserver(ViEEncoder* owner)
: owner_(owner) {
}
virtual ~ViEBitrateObserver() {}
// Implements BitrateObserver.
virtual void OnNetworkChanged(const uint32_t bitrate_bps,
const uint8_t fraction_lost,
const uint32_t rtt) {
owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
}
private:
ViEEncoder* owner_;
};
class ViEPacedSenderCallback : public PacedSender::Callback {
public:
explicit ViEPacedSenderCallback(ViEEncoder* owner)
: owner_(owner) {
}
virtual ~ViEPacedSenderCallback() {}
virtual bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission) {
return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
retransmission);
}
virtual int TimeToSendPadding(int bytes) {
return owner_->TimeToSendPadding(bytes);
}
private:
ViEEncoder* owner_;
};
ViEEncoder::ViEEncoder(int32_t engine_id,
int32_t channel_id,
uint32_t number_of_cores,
const Config& config,
ProcessThread& module_process_thread,
BitrateController* bitrate_controller)
: engine_id_(engine_id),
channel_id_(channel_id),
number_of_cores_(number_of_cores),
vcm_(*webrtc::VideoCodingModule::Create()),
vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
channel_id))),
callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
bitrate_controller_(bitrate_controller),
time_of_last_incoming_frame_ms_(0),
send_padding_(false),
min_transmit_bitrate_kbps_(0),
target_delay_ms_(0),
network_is_transmitting_(true),
encoder_paused_(false),
encoder_paused_and_dropped_frame_(false),
fec_enabled_(false),
nack_enabled_(false),
codec_observer_(NULL),
effect_filter_(NULL),
module_process_thread_(module_process_thread),
has_received_sli_(false),
picture_id_sli_(0),
has_received_rpsi_(false),
picture_id_rpsi_(0),
qm_callback_(NULL),
video_suspended_(false),
pre_encode_callback_(NULL),
start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
RtpRtcp::Configuration configuration;
configuration.id = ViEModuleId(engine_id_, channel_id_);
configuration.audio = false; // Video.
default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
bitrate_observer_.reset(new ViEBitrateObserver(this));
pacing_callback_.reset(new ViEPacedSenderCallback(this));
paced_sender_.reset(new PacedSender(
Clock::GetRealTimeClock(),
pacing_callback_.get(),
kDefaultStartBitrateKbps,
PacedSender::kDefaultPaceMultiplier * kDefaultStartBitrateKbps,
0));
}
bool ViEEncoder::Init() {
if (vcm_.InitializeSender() != 0) {
return false;
}
vpm_.EnableTemporalDecimation(true);
// Enable/disable content analysis: off by default for now.
vpm_.EnableContentAnalysis(false);
if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
return false;
}
if (qm_callback_) {
delete qm_callback_;
}
qm_callback_ = new QMVideoSettingsCallback(&vpm_);
#ifdef VIDEOCODEC_VP8
VideoCodecType codec_type = webrtc::kVideoCodecVP8;
#else
VideoCodecType codec_type = webrtc::kVideoCodecI420;
#endif
VideoCodec video_codec;
if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
return false;
}
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
return false;
}
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
return false;
}
if (vcm_.RegisterTransportCallback(this) != 0) {
return false;
}
if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
return false;
}
if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
return false;
}
return true;
}
ViEEncoder::~ViEEncoder() {
UpdateHistograms();
if (bitrate_controller_) {
bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
}
module_process_thread_.DeRegisterModule(&vcm_);
module_process_thread_.DeRegisterModule(&vpm_);
module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
module_process_thread_.DeRegisterModule(paced_sender_.get());
VideoCodingModule::Destroy(&vcm_);
VideoProcessingModule::Destroy(&vpm_);
delete qm_callback_;
}
void ViEEncoder::UpdateHistograms() {
const float kMinCallLengthInMinutes = 0.5f;
float elapsed_minutes =
(Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
if (elapsed_minutes < kMinCallLengthInMinutes) {
return;
}
webrtc::VCMFrameCount frames;
if (vcm_.SentFrameCount(frames) != VCM_OK) {
return;
}
uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
if (total_frames > 0) {
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
static_cast<int>(
(frames.numKeyFrames * 1000.0f / total_frames) + 0.5f));
}
}
int ViEEncoder::Owner() const {
return channel_id_;
}
void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
{
CriticalSectionScoped cs(data_cs_.get());
network_is_transmitting_ = is_transmitting;
}
if (is_transmitting) {
paced_sender_->Resume();
} else {
paced_sender_->Pause();
}
}
void ViEEncoder::Pause() {
CriticalSectionScoped cs(data_cs_.get());
encoder_paused_ = true;
}
void ViEEncoder::Restart() {
CriticalSectionScoped cs(data_cs_.get());
encoder_paused_ = false;
}
uint8_t ViEEncoder::NumberOfCodecs() {
return vcm_.NumberOfCodecs();
}
int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
if (vcm_.Codec(list_index, video_codec) != 0) {
return -1;
}
return 0;
}
int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
uint8_t pl_type,
bool internal_source) {
if (encoder == NULL)
return -1;
if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
VCM_OK) {
return -1;
}
return 0;
}
int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
webrtc::VideoCodec current_send_codec;
if (vcm_.SendCodec(&current_send_codec) == VCM_OK) {
uint32_t current_bitrate_bps = 0;
if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
}
current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
}
if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
return -1;
}
// If the external encoder is the current send codec, use vcm internal
// encoder.
if (current_send_codec.plType == pl_type) {
uint16_t max_data_payload_length =
default_rtp_rtcp_->MaxDataPayloadLength();
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
}
// TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
// raw pointer to an |extra_options| that's long gone. Clearing it here is
// a hack to prevent the following code from crashing. This should be fixed
// for realz. https://code.google.com/p/chromium/issues/detail?id=348222
current_send_codec.extra_options = NULL;
if (vcm_.RegisterSendCodec(&current_send_codec, number_of_cores_,
max_data_payload_length) != VCM_OK) {
LOG(LS_INFO) << "De-registered the currently used external encoder ("
<< static_cast<int>(pl_type) << ") and therefore tried to "
<< "register the corresponding internal encoder, but none "
<< "was supported.";
}
}
return 0;
}
int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
// Setting target width and height for VPM.
if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
video_codec.maxFramerate) != VPM_OK) {
return -1;
}
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
return -1;
}
// Convert from kbps to bps.
std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
video_codec.startBitrate * 1000,
video_codec.simulcastStream,
video_codec.numberOfSimulcastStreams);
default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
uint16_t max_data_payload_length =
default_rtp_rtcp_->MaxDataPayloadLength();
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
max_data_payload_length) != VCM_OK) {
return -1;
}
// Set this module as sending right away, let the slave module in the channel
// start and stop sending.
if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
return -1;
}
bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
video_codec.startBitrate * 1000,
video_codec.minBitrate * 1000,
kTransmissionMaxBitrateMultiplier *
video_codec.maxBitrate * 1000);
CriticalSectionScoped crit(data_cs_.get());
int pad_up_to_bitrate_kbps = video_codec.startBitrate;
if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
paced_sender_->UpdateBitrate(
video_codec.startBitrate,
PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate,
pad_up_to_bitrate_kbps);
return 0;
}
int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
if (vcm_.SendCodec(video_codec) != 0) {
return -1;
}
return 0;
}
int32_t ViEEncoder::GetCodecConfigParameters(
unsigned char config_parameters[kConfigParameterSize],
unsigned char& config_parameters_size) {
int32_t num_parameters =
vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
if (num_parameters <= 0) {
config_parameters_size = 0;
return -1;
}
config_parameters_size = static_cast<unsigned char>(num_parameters);
return 0;
}
int32_t ViEEncoder::ScaleInputImage(bool enable) {
VideoFrameResampling resampling_mode = kFastRescaling;
// TODO(mflodman) What?
if (enable) {
// kInterpolation is currently not supported.
LOG_F(LS_ERROR) << "Not supported.";
return -1;
}
vpm_.SetInputFrameResampleMode(resampling_mode);
return 0;
}
bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission) {
return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
capture_time_ms, retransmission);
}
int ViEEncoder::TimeToSendPadding(int bytes) {
bool send_padding;
{
CriticalSectionScoped cs(data_cs_.get());
send_padding =
send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
}
if (send_padding) {
return default_rtp_rtcp_->TimeToSendPadding(bytes);
}
return 0;
}
bool ViEEncoder::EncoderPaused() const {
// Pause video if paused by caller or as long as the network is down or the
// pacer queue has grown too large in buffered mode.
if (encoder_paused_) {
return true;
}
if (target_delay_ms_ > 0) {
// Buffered mode.
// TODO(pwestin): Workaround until nack is configured as a time and not
// number of packets.
return paced_sender_->QueueInMs() >=
std::max(static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
kMinPacingDelayMs);
}
if (paced_sender_->ExpectedQueueTimeMs() >
PacedSender::kDefaultMaxQueueLengthMs) {
// Too much data in pacer queue, drop frame.
return true;
}
return !network_is_transmitting_;
}
void ViEEncoder::TraceFrameDropStart() {
// Start trace event only on the first frame after encoder is paused.
if (!encoder_paused_and_dropped_frame_) {
TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
}
encoder_paused_and_dropped_frame_ = true;
return;
}
void ViEEncoder::TraceFrameDropEnd() {
// End trace event on first frame after encoder resumes, if frame was dropped.
if (encoder_paused_and_dropped_frame_) {
TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
}
encoder_paused_and_dropped_frame_ = false;
}
RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
return default_rtp_rtcp_.get();
}
void ViEEncoder::DeliverFrame(int id,
I420VideoFrame* video_frame,
int num_csrcs,
const uint32_t CSRC[kRtpCsrcSize]) {
if (default_rtp_rtcp_->SendingMedia() == false) {
// We've paused or we have no channels attached, don't encode.
return;
}
{
CriticalSectionScoped cs(data_cs_.get());
time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
if (EncoderPaused()) {
TraceFrameDropStart();
return;
}
TraceFrameDropEnd();
}
// Convert render time, in ms, to RTP timestamp.
const int kMsToRtpTimestamp = 90;
const uint32_t time_stamp =
kMsToRtpTimestamp *
static_cast<uint32_t>(video_frame->render_time_ms());
TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
"Encode");
video_frame->set_timestamp(time_stamp);
// Make sure the CSRC list is correct.
if (num_csrcs > 0) {
uint32_t tempCSRC[kRtpCsrcSize];
for (int i = 0; i < num_csrcs; i++) {
if (CSRC[i] == 1) {
tempCSRC[i] = default_rtp_rtcp_->SSRC();
} else {
tempCSRC[i] = CSRC[i];
}
}
default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
}
I420VideoFrame* decimated_frame = NULL;
// TODO(wuchengli): support texture frames.
if (video_frame->native_handle() == NULL) {
{
CriticalSectionScoped cs(callback_cs_.get());
if (effect_filter_) {
unsigned int length =
CalcBufferSize(kI420, video_frame->width(), video_frame->height());
scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
ExtractBuffer(*video_frame, length, video_buffer.get());
effect_filter_->Transform(length,
video_buffer.get(),
video_frame->ntp_time_ms(),
video_frame->timestamp(),
video_frame->width(),
video_frame->height());
}
}
// Pass frame via preprocessor.
const int ret = vpm_.PreprocessFrame(*video_frame, &decimated_frame);
if (ret == 1) {
// Drop this frame.
return;
}
if (ret != VPM_OK) {
return;
}
}
// If the frame was not resampled or scaled => use original.
if (decimated_frame == NULL) {
decimated_frame = video_frame;
}
{
CriticalSectionScoped cs(callback_cs_.get());
if (pre_encode_callback_)
pre_encode_callback_->FrameCallback(decimated_frame);
}
if (video_frame->native_handle() != NULL) {
// TODO(wuchengli): add texture support. http://crbug.com/362437
return;
}
#ifdef VIDEOCODEC_VP8
if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
webrtc::CodecSpecificInfo codec_specific_info;
codec_specific_info.codecType = webrtc::kVideoCodecVP8;
{
CriticalSectionScoped cs(data_cs_.get());
codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
has_received_rpsi_;
codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
has_received_sli_;
codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
picture_id_rpsi_;
codec_specific_info.codecSpecific.VP8.pictureIdSLI =
picture_id_sli_;
has_received_sli_ = false;
has_received_rpsi_ = false;
}
vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics(),
&codec_specific_info);
return;
}
#endif
vcm_.AddVideoFrame(*decimated_frame);
}
void ViEEncoder::DelayChanged(int id, int frame_delay) {
default_rtp_rtcp_->SetCameraDelay(frame_delay);
}
int ViEEncoder::GetPreferedFrameSettings(int* width,
int* height,
int* frame_rate) {
webrtc::VideoCodec video_codec;
memset(&video_codec, 0, sizeof(video_codec));
if (vcm_.SendCodec(&video_codec) != VCM_OK) {
return -1;
}
*width = video_codec.width;
*height = video_codec.height;
*frame_rate = video_codec.maxFramerate;
return 0;
}
int ViEEncoder::SendKeyFrame() {
return vcm_.IntraFrameRequest(0);
}
int32_t ViEEncoder::SendCodecStatistics(
uint32_t* num_key_frames, uint32_t* num_delta_frames) {
webrtc::VCMFrameCount sent_frames;
if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
return -1;
}
*num_key_frames = sent_frames.numKeyFrames;
*num_delta_frames = sent_frames.numDeltaFrames;
return 0;
}
int32_t ViEEncoder::PacerQueuingDelayMs() const {
return paced_sender_->QueueInMs();
}
int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
if (vcm_.Bitrate(bitrate) != 0)
return -1;
return 0;
}
int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
bool fec_enabled = false;
uint8_t dummy_ptype_red = 0;
uint8_t dummy_ptypeFEC = 0;
// Updated protection method to VCM to get correct packetization sizes.
// FEC has larger overhead than NACK -> set FEC if used.
int32_t error = default_rtp_rtcp_->GenericFECStatus(fec_enabled,
dummy_ptype_red,
dummy_ptypeFEC);
if (error) {
return -1;
}
if (fec_enabled_ == fec_enabled && nack_enabled_ == enable_nack) {
// No change needed, we're already in correct state.
return 0;
}
fec_enabled_ = fec_enabled;
nack_enabled_ = enable_nack;
// Set Video Protection for VCM.
if (fec_enabled && nack_enabled_) {
vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
} else {
vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
vcm_.SetVideoProtection(webrtc::kProtectionNackSender, nack_enabled_);
vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
}
if (fec_enabled_ || nack_enabled_) {
vcm_.RegisterProtectionCallback(this);
// The send codec must be registered to set correct MTU.
webrtc::VideoCodec codec;
if (vcm_.SendCodec(&codec) == 0) {
uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
uint32_t current_bitrate_bps = 0;
if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
LOG_F(LS_WARNING) <<
"Failed to get the current encoder target bitrate.";
}
// Convert to start bitrate in kbps.
codec.startBitrate = (current_bitrate_bps + 500) / 1000;
if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
return -1;
}
}
return 0;
} else {
// FEC and NACK are disabled.
vcm_.RegisterProtectionCallback(NULL);
}
return 0;
}
void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
{
CriticalSectionScoped cs(data_cs_.get());
target_delay_ms_ = target_delay_ms;
}
if (target_delay_ms > 0) {
// Disable external frame-droppers.
vcm_.EnableFrameDropper(false);
vpm_.EnableTemporalDecimation(false);
} else {
// Real-time mode - enable frame droppers.
vpm_.EnableTemporalDecimation(true);
vcm_.EnableFrameDropper(true);
}
}
int32_t ViEEncoder::SendData(
const FrameType frame_type,
const uint8_t payload_type,
const uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
const uint32_t payload_size,
const webrtc::RTPFragmentationHeader& fragmentation_header,
const RTPVideoHeader* rtp_video_hdr) {
// New encoded data, hand over to the rtp module.
return default_rtp_rtcp_->SendOutgoingData(frame_type,
payload_type,
time_stamp,
capture_time_ms,
payload_data,
payload_size,
&fragmentation_header,
rtp_video_hdr);
}
int32_t ViEEncoder::ProtectionRequest(
const FecProtectionParams* delta_fec_params,
const FecProtectionParams* key_fec_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) {
default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
sent_nack_rate_bps);
return 0;
}
int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
const uint32_t frame_rate) {
CriticalSectionScoped cs(callback_cs_.get());
if (codec_observer_) {
codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
}
return 0;
}
int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
CriticalSectionScoped cs(callback_cs_.get());
if (observer && codec_observer_) {
LOG_F(LS_ERROR) << "Observer already set.";
return -1;
}
codec_observer_ = observer;
return 0;
}
void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
uint8_t picture_id) {
CriticalSectionScoped cs(data_cs_.get());
picture_id_sli_ = picture_id;
has_received_sli_ = true;
}
void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
uint64_t picture_id) {
CriticalSectionScoped cs(data_cs_.get());
picture_id_rpsi_ = picture_id;
has_received_rpsi_ = true;
}
void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
// Key frame request from remote side, signal to VCM.
TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
int idx = 0;
{
CriticalSectionScoped cs(data_cs_.get());
std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
if (stream_it == ssrc_streams_.end()) {
LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
<< ssrc_streams_.size();
return;
}
std::map<unsigned int, int64_t>::iterator time_it =
time_last_intra_request_ms_.find(ssrc);
if (time_it == time_last_intra_request_ms_.end()) {
time_last_intra_request_ms_[ssrc] = 0;
}
int64_t now = TickTime::MillisecondTimestamp();
if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
return;
}
time_last_intra_request_ms_[ssrc] = now;
idx = stream_it->second;
}
// Release the critsect before triggering key frame.
vcm_.IntraFrameRequest(idx);
}
void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
CriticalSectionScoped cs(data_cs_.get());
std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
if (it == ssrc_streams_.end()) {
return;
}
ssrc_streams_[new_ssrc] = it->second;
ssrc_streams_.erase(it);
std::map<unsigned int, int64_t>::iterator time_it =
time_last_intra_request_ms_.find(old_ssrc);
int64_t last_intra_request_ms = 0;
if (time_it != time_last_intra_request_ms_.end()) {
last_intra_request_ms = time_it->second;
time_last_intra_request_ms_.erase(time_it);
}
time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
}
bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
VideoCodec codec;
if (vcm_.SendCodec(&codec) != 0)
return false;
if (codec.numberOfSimulcastStreams > 0 &&
ssrcs.size() != codec.numberOfSimulcastStreams) {
return false;
}
CriticalSectionScoped cs(data_cs_.get());
ssrc_streams_.clear();
time_last_intra_request_ms_.clear();
int idx = 0;
for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
it != ssrcs.end(); ++it, ++idx) {
unsigned int ssrc = *it;
ssrc_streams_[ssrc] = idx;
}
return true;
}
void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
assert(min_transmit_bitrate_kbps >= 0);
CriticalSectionScoped crit(data_cs_.get());
min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
}
// Called from ViEBitrateObserver.
void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
const uint8_t fraction_lost,
const uint32_t round_trip_time_ms) {
LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
<< " packet loss " << fraction_lost
<< " rtt " << round_trip_time_ms;
vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
bool video_is_suspended = vcm_.VideoSuspended();
int bitrate_kbps = bitrate_bps / 1000;
VideoCodec send_codec;
if (vcm_.SendCodec(&send_codec) != 0) {
return;
}
SimulcastStream* stream_configs = send_codec.simulcastStream;
// Allocate the bandwidth between the streams.
std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
bitrate_bps,
stream_configs,
send_codec.numberOfSimulcastStreams);
// Find the max amount of padding we can allow ourselves to send at this
// point, based on which streams are currently active and what our current
// available bandwidth is.
int pad_up_to_bitrate_kbps = 0;
if (send_codec.numberOfSimulcastStreams == 0) {
pad_up_to_bitrate_kbps = send_codec.minBitrate;
} else {
pad_up_to_bitrate_kbps =
stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
}
}
// Disable padding if only sending one stream and video isn't suspended and
// min-transmit bitrate isn't used (applied later).
if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
pad_up_to_bitrate_kbps = 0;
{
CriticalSectionScoped cs(data_cs_.get());
// The amount of padding should decay to zero if no frames are being
// captured unless a min-transmit bitrate is used.
int64_t now_ms = TickTime::MillisecondTimestamp();
if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
pad_up_to_bitrate_kbps = 0;
// Pad up to min bitrate.
if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
// Padding may never exceed bitrate estimate.
if (pad_up_to_bitrate_kbps > bitrate_kbps)
pad_up_to_bitrate_kbps = bitrate_kbps;
paced_sender_->UpdateBitrate(
bitrate_kbps,
PacedSender::kDefaultPaceMultiplier * bitrate_kbps,
pad_up_to_bitrate_kbps);
default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
if (video_suspended_ == video_is_suspended)
return;
video_suspended_ = video_is_suspended;
}
// Video suspend-state changed, inform codec observer.
CriticalSectionScoped crit(callback_cs_.get());
if (codec_observer_) {
LOG(LS_INFO) << "Video suspended " << video_is_suspended
<< " for channel " << channel_id_;
codec_observer_->SuspendChange(channel_id_, video_is_suspended);
}
}
PacedSender* ViEEncoder::GetPacedSender() {
return paced_sender_.get();
}
int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
CriticalSectionScoped cs(callback_cs_.get());
if (effect_filter != NULL && effect_filter_ != NULL) {
LOG_F(LS_ERROR) << "Filter already set.";
return -1;
}
effect_filter_ = effect_filter;
return 0;
}
int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
return vcm_.StartDebugRecording(fileNameUTF8);
}
int ViEEncoder::StopDebugRecording() {
return vcm_.StopDebugRecording();
}
void ViEEncoder::SuspendBelowMinBitrate() {
vcm_.SuspendBelowMinBitrate();
bitrate_controller_->EnforceMinBitrate(false);
}
void ViEEncoder::RegisterPreEncodeCallback(
I420FrameCallback* pre_encode_callback) {
CriticalSectionScoped cs(callback_cs_.get());
pre_encode_callback_ = pre_encode_callback;
}
void ViEEncoder::DeRegisterPreEncodeCallback() {
CriticalSectionScoped cs(callback_cs_.get());
pre_encode_callback_ = NULL;
}
void ViEEncoder::RegisterPostEncodeImageCallback(
EncodedImageCallback* post_encode_callback) {
vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
}
void ViEEncoder::DeRegisterPostEncodeImageCallback() {
vcm_.RegisterPostEncodeImageCallback(NULL);
}
QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
: vpm_(vpm) {
}
QMVideoSettingsCallback::~QMVideoSettingsCallback() {
}
int32_t QMVideoSettingsCallback::SetVideoQMSettings(
const uint32_t frame_rate,
const uint32_t width,
const uint32_t height) {
return vpm_->SetTargetResolution(width, height, frame_rate);
}
} // namespace webrtc