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5e161616b17900c06809e7275afca96363d44ad5
platform-external-webrtc/talk/media/webrtc/webrtcvideoengine2.cc
pbos@webrtc.org 5e161616b1 Remove CPU monitor from WebRtcVideoEngine2. 
CPU adaptation is based on timings done inside webrtc, not actual CPU
values anymore. This code has never been wired up and is causing flakes
on at least valgrind, but possibly also on actual platforms.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34089004

Cr-Commit-Position: refs/heads/master@{#8221}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8221 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 15:31:26 +00:00

2153 lines
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/*
* libjingle
* Copyright 2014 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_WEBRTC_VIDEO
#include "talk/media/webrtc/webrtcvideoengine2.h"
#include <algorithm>
#include <set>
#include <string>
#include "libyuv/convert_from.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videorenderer.h"
#include "talk/media/webrtc/constants.h"
#include "talk/media/webrtc/simulcast.h"
#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "talk/media/webrtc/webrtcvideoengine.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
#define UNIMPLEMENTED \
LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
ASSERT(false)
namespace cricket {
namespace {
static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
std::stringstream out;
out << '{';
for (size_t i = 0; i < codecs.size(); ++i) {
out << codecs[i].ToString();
if (i != codecs.size() - 1) {
out << ", ";
}
}
out << '}';
return out.str();
}
static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) {
bool has_video = false;
for (size_t i = 0; i < codecs.size(); ++i) {
if (!codecs[i].ValidateCodecFormat()) {
return false;
}
if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) {
has_video = true;
}
}
if (!has_video) {
LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: "
<< CodecVectorToString(codecs);
return false;
}
return true;
}
static std::string RtpExtensionsToString(
const std::vector<RtpHeaderExtension>& extensions) {
std::stringstream out;
out << '{';
for (size_t i = 0; i < extensions.size(); ++i) {
out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
if (i != extensions.size() - 1) {
out << ", ";
}
}
out << '}';
return out.str();
}
// Merges two fec configs and logs an error if a conflict arises
// such that merging in diferent order would trigger a diferent output.
static void MergeFecConfig(const webrtc::FecConfig& other,
webrtc::FecConfig* output) {
if (other.ulpfec_payload_type != -1) {
if (output->ulpfec_payload_type != -1 &&
output->ulpfec_payload_type != other.ulpfec_payload_type) {
LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
<< output->ulpfec_payload_type << " and "
<< other.ulpfec_payload_type;
}
output->ulpfec_payload_type = other.ulpfec_payload_type;
}
if (other.red_payload_type != -1) {
if (output->red_payload_type != -1 &&
output->red_payload_type != other.red_payload_type) {
LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
<< output->red_payload_type << " and "
<< other.red_payload_type;
}
output->red_payload_type = other.red_payload_type;
}
}
} // namespace
// This constant is really an on/off, lower-level configurable NACK history
// duration hasn't been implemented.
static const int kNackHistoryMs = 1000;
static const int kDefaultQpMax = 56;
static const int kDefaultRtcpReceiverReportSsrc = 1;
// External video encoders are given payloads 120-127. This also means that we
// only support up to 8 external payload types.
static const int kExternalVideoPayloadTypeBase = 120;
#ifndef NDEBUG
static const size_t kMaxExternalVideoCodecs = 8;
#endif
const char kH264CodecName[] = "H264";
static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs,
const VideoCodec& requested_codec,
VideoCodec* matching_codec) {
for (size_t i = 0; i < codecs.size(); ++i) {
if (requested_codec.Matches(codecs[i])) {
*matching_codec = codecs[i];
return true;
}
}
return false;
}
static bool ValidateRtpHeaderExtensionIds(
const std::vector<RtpHeaderExtension>& extensions) {
std::set<int> extensions_used;
for (size_t i = 0; i < extensions.size(); ++i) {
if (extensions[i].id < 0 || extensions[i].id >= 15 ||
!extensions_used.insert(extensions[i].id).second) {
LOG(LS_ERROR) << "RTP extensions are with incorrect or duplicate ids.";
return false;
}
}
return true;
}
static bool CompareRtpHeaderExtensionIds(
const webrtc::RtpExtension& extension1,
const webrtc::RtpExtension& extension2) {
// Sorting on ID is sufficient, more than one extension per ID is unsupported.
return extension1.id > extension2.id;
}
static std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
std::vector<webrtc::RtpExtension> webrtc_extensions;
for (size_t i = 0; i < extensions.size(); ++i) {
// Unsupported extensions will be ignored.
if (webrtc::RtpExtension::IsSupported(extensions[i].uri)) {
webrtc_extensions.push_back(webrtc::RtpExtension(
extensions[i].uri, extensions[i].id));
} else {
LOG(LS_WARNING) << "Unsupported RTP extension: " << extensions[i].uri;
}
}
// Sort filtered headers to make sure that they can later be compared
// regardless of in which order they were entered.
std::sort(webrtc_extensions.begin(), webrtc_extensions.end(),
CompareRtpHeaderExtensionIds);
return webrtc_extensions;
}
static bool RtpExtensionsHaveChanged(
const std::vector<webrtc::RtpExtension>& before,
const std::vector<webrtc::RtpExtension>& after) {
if (before.size() != after.size())
return true;
for (size_t i = 0; i < before.size(); ++i) {
if (before[i].id != after[i].id)
return true;
if (before[i].name != after[i].name)
return true;
}
return false;
}
std::vector<webrtc::VideoStream>
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
const VideoCodec& codec,
const VideoOptions& options,
size_t num_streams) {
// Use default factory for non-simulcast.
int max_qp = kDefaultQpMax;
codec.GetParam(kCodecParamMaxQuantization, &max_qp);
int min_bitrate_kbps;
if (!codec.GetParam(kCodecParamMinBitrate, &min_bitrate_kbps) ||
min_bitrate_kbps < kMinVideoBitrate) {
min_bitrate_kbps = kMinVideoBitrate;
}
int max_bitrate_kbps;
if (!codec.GetParam(kCodecParamMaxBitrate, &max_bitrate_kbps)) {
max_bitrate_kbps = 0;
}
return GetSimulcastConfig(
num_streams,
GetSimulcastBitrateMode(options),
codec.width,
codec.height,
max_bitrate_kbps * 1000,
max_qp,
codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate);
}
std::vector<webrtc::VideoStream>
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoStreams(
const VideoCodec& codec,
const VideoOptions& options,
size_t num_streams) {
if (num_streams != 1)
return CreateSimulcastVideoStreams(codec, options, num_streams);
webrtc::VideoStream stream;
stream.width = codec.width;
stream.height = codec.height;
stream.max_framerate =
codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
stream.min_bitrate_bps = kMinVideoBitrate * 1000;
stream.target_bitrate_bps = stream.max_bitrate_bps = kMaxVideoBitrate * 1000;
int max_qp = kDefaultQpMax;
codec.GetParam(kCodecParamMaxQuantization, &max_qp);
stream.max_qp = max_qp;
std::vector<webrtc::VideoStream> streams;
streams.push_back(stream);
return streams;
}
void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
const VideoCodec& codec,
const VideoOptions& options) {
if (CodecNameMatches(codec.name, kVp8CodecName)) {
encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
options.video_noise_reduction.Get(&encoder_settings_.vp8.denoisingOn);
return &encoder_settings_.vp8;
}
if (CodecNameMatches(codec.name, kVp9CodecName)) {
encoder_settings_.vp9 = webrtc::VideoEncoder::GetDefaultVp9Settings();
options.video_noise_reduction.Get(&encoder_settings_.vp9.denoisingOn);
return &encoder_settings_.vp9;
}
return NULL;
}
DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
: default_recv_ssrc_(0), default_renderer_(NULL) {}
UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
VideoMediaChannel* channel,
uint32_t ssrc) {
if (default_recv_ssrc_ != 0) { // Already one default stream.
LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set.";
return kDropPacket;
}
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
if (!channel->AddRecvStream(sp)) {
LOG(LS_WARNING) << "Could not create default receive stream.";
}
channel->SetRenderer(ssrc, default_renderer_);
default_recv_ssrc_ = ssrc;
return kDeliverPacket;
}
WebRtcCallFactory::~WebRtcCallFactory() {
}
webrtc::Call* WebRtcCallFactory::CreateCall(
const webrtc::Call::Config& config) {
return webrtc::Call::Create(config);
}
VideoRenderer* DefaultUnsignalledSsrcHandler::GetDefaultRenderer() const {
return default_renderer_;
}
void DefaultUnsignalledSsrcHandler::SetDefaultRenderer(
VideoMediaChannel* channel,
VideoRenderer* renderer) {
default_renderer_ = renderer;
if (default_recv_ssrc_ != 0) {
channel->SetRenderer(default_recv_ssrc_, default_renderer_);
}
}
WebRtcVideoEngine2::WebRtcVideoEngine2()
: worker_thread_(NULL),
voice_engine_(NULL),
default_codec_format_(kDefaultVideoMaxWidth,
kDefaultVideoMaxHeight,
FPS_TO_INTERVAL(kDefaultVideoMaxFramerate),
FOURCC_ANY),
initialized_(false),
call_factory_(&default_call_factory_),
external_decoder_factory_(NULL),
external_encoder_factory_(NULL) {
LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
video_codecs_ = GetSupportedCodecs();
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
kRtpTimestampOffsetHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
if (initialized_) {
Terminate();
}
}
void WebRtcVideoEngine2::SetCallFactory(WebRtcCallFactory* call_factory) {
assert(!initialized_);
call_factory_ = call_factory;
}
bool WebRtcVideoEngine2::Init(rtc::Thread* worker_thread) {
LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
worker_thread_ = worker_thread;
ASSERT(worker_thread_ != NULL);
initialized_ = true;
return true;
}
void WebRtcVideoEngine2::Terminate() {
LOG(LS_INFO) << "WebRtcVideoEngine2::Terminate";
initialized_ = false;
}
int WebRtcVideoEngine2::GetCapabilities() { return VIDEO_RECV | VIDEO_SEND; }
bool WebRtcVideoEngine2::SetDefaultEncoderConfig(
const VideoEncoderConfig& config) {
const VideoCodec& codec = config.max_codec;
bool supports_codec = false;
for (size_t i = 0; i < video_codecs_.size(); ++i) {
if (CodecNameMatches(video_codecs_[i].name, codec.name)) {
video_codecs_[i] = codec;
supports_codec = true;
break;
}
}
if (!supports_codec) {
LOG(LS_ERROR) << "SetDefaultEncoderConfig, codec not supported: "
<< codec.ToString();
return false;
}
default_codec_format_ =
VideoFormat(codec.width,
codec.height,
VideoFormat::FpsToInterval(codec.framerate),
FOURCC_ANY);
return true;
}
WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
const VideoOptions& options,
VoiceMediaChannel* voice_channel) {
assert(initialized_);
LOG(LS_INFO) << "CreateChannel: "
<< (voice_channel != NULL ? "With" : "Without")
<< " voice channel. Options: " << options.ToString();
WebRtcVideoChannel2* channel =
new WebRtcVideoChannel2(call_factory_,
voice_engine_,
voice_channel,
options,
external_encoder_factory_,
external_decoder_factory_);
if (!channel->Init()) {
delete channel;
return NULL;
}
channel->SetRecvCodecs(video_codecs_);
return channel;
}
const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
return video_codecs_;
}
const std::vector<RtpHeaderExtension>&
WebRtcVideoEngine2::rtp_header_extensions() const {
return rtp_header_extensions_;
}
void WebRtcVideoEngine2::SetLogging(int min_sev, const char* filter) {
// TODO(pbos): Set up logging.
LOG(LS_VERBOSE) << "SetLogging: " << min_sev << '"' << filter << '"';
// if min_sev == -1, we keep the current log level.
if (min_sev < 0) {
assert(min_sev == -1);
return;
}
}
void WebRtcVideoEngine2::SetExternalDecoderFactory(
WebRtcVideoDecoderFactory* decoder_factory) {
assert(!initialized_);
external_decoder_factory_ = decoder_factory;
}
void WebRtcVideoEngine2::SetExternalEncoderFactory(
WebRtcVideoEncoderFactory* encoder_factory) {
assert(!initialized_);
if (external_encoder_factory_ == encoder_factory)
return;
// No matter what happens we shouldn't hold on to a stale
// WebRtcSimulcastEncoderFactory.
simulcast_encoder_factory_.reset();
if (encoder_factory &&
WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory(
encoder_factory->codecs())) {
simulcast_encoder_factory_.reset(
new WebRtcSimulcastEncoderFactory(encoder_factory));
encoder_factory = simulcast_encoder_factory_.get();
}
external_encoder_factory_ = encoder_factory;
video_codecs_ = GetSupportedCodecs();
}
bool WebRtcVideoEngine2::EnableTimedRender() {
// TODO(pbos): Figure out whether this can be removed.
return true;
}
// Checks to see whether we comprehend and could receive a particular codec
bool WebRtcVideoEngine2::FindCodec(const VideoCodec& in) {
// TODO(pbos): Probe encoder factory to figure out that the codec is supported
// if supported by the encoder factory. Add a corresponding test that fails
// with this code (that doesn't ask the factory).
for (size_t j = 0; j < video_codecs_.size(); ++j) {
VideoCodec codec(video_codecs_[j].id, video_codecs_[j].name, 0, 0, 0, 0);
if (codec.Matches(in)) {
return true;
}
}
return false;
}
// Tells whether the |requested| codec can be transmitted or not. If it can be
// transmitted |out| is set with the best settings supported. Aspect ratio will
// be set as close to |current|'s as possible. If not set |requested|'s
// dimensions will be used for aspect ratio matching.
bool WebRtcVideoEngine2::CanSendCodec(const VideoCodec& requested,
const VideoCodec& current,
VideoCodec* out) {
assert(out != NULL);
if (requested.width != requested.height &&
(requested.height == 0 || requested.width == 0)) {
// 0xn and nx0 are invalid resolutions.
return false;
}
VideoCodec matching_codec;
if (!FindFirstMatchingCodec(video_codecs_, requested, &matching_codec)) {
// Codec not supported.
return false;
}
out->id = requested.id;
out->name = requested.name;
out->preference = requested.preference;
out->params = requested.params;
out->framerate =
rtc::_min(requested.framerate, matching_codec.framerate);
out->params = requested.params;
out->feedback_params = requested.feedback_params;
out->width = requested.width;
out->height = requested.height;
if (requested.width == 0 && requested.height == 0) {
return true;
}
while (out->width > matching_codec.width) {
out->width /= 2;
out->height /= 2;
}
return out->width > 0 && out->height > 0;
}
bool WebRtcVideoEngine2::SetVoiceEngine(WebRtcVoiceEngine* voice_engine) {
if (initialized_) {
LOG(LS_WARNING) << "SetVoiceEngine can not be called after Init";
return false;
}
voice_engine_ = voice_engine;
return true;
}
// Ignore spammy trace messages, mostly from the stats API when we haven't
// gotten RTCP info yet from the remote side.
bool WebRtcVideoEngine2::ShouldIgnoreTrace(const std::string& trace) {
static const char* const kTracesToIgnore[] = {NULL};
for (const char* const* p = kTracesToIgnore; *p; ++p) {
if (trace.find(*p) == 0) {
return true;
}
}
return false;
}
std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
if (external_encoder_factory_ == NULL) {
return supported_codecs;
}
assert(external_encoder_factory_->codecs().size() <= kMaxExternalVideoCodecs);
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
external_encoder_factory_->codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
// Don't add internally-supported codecs twice.
if (CodecIsInternallySupported(codecs[i].name)) {
continue;
}
VideoCodec codec(kExternalVideoPayloadTypeBase + static_cast<int>(i),
codecs[i].name,
codecs[i].max_width,
codecs[i].max_height,
codecs[i].max_fps,
0);
AddDefaultFeedbackParams(&codec);
supported_codecs.push_back(codec);
}
return supported_codecs;
}
WebRtcVideoChannel2::WebRtcVideoChannel2(
WebRtcCallFactory* call_factory,
WebRtcVoiceEngine* voice_engine,
VoiceMediaChannel* voice_channel,
const VideoOptions& options,
WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory)
: unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
voice_channel_(voice_channel),
external_encoder_factory_(external_encoder_factory),
external_decoder_factory_(external_decoder_factory) {
SetDefaultOptions();
options_.SetAll(options);
webrtc::Call::Config config(this);
config.overuse_callback = this;
if (voice_engine != NULL) {
config.voice_engine = voice_engine->voe()->engine();
}
call_.reset(call_factory->CreateCall(config));
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
default_send_ssrc_ = 0;
}
void WebRtcVideoChannel2::SetDefaultOptions() {
options_.cpu_overuse_detection.Set(false);
options_.dscp.Set(false);
options_.suspend_below_min_bitrate.Set(false);
options_.video_noise_reduction.Set(true);
options_.screencast_min_bitrate.Set(0);
}
WebRtcVideoChannel2::~WebRtcVideoChannel2() {
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
delete it->second;
}
for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
it != receive_streams_.end();
++it) {
delete it->second;
}
}
bool WebRtcVideoChannel2::Init() { return true; }
bool WebRtcVideoChannel2::CodecIsExternallySupported(
const std::string& name) const {
if (external_encoder_factory_ == NULL) {
return false;
}
const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
external_encoder_factory_->codecs();
for (size_t c = 0; c < external_codecs.size(); ++c) {
if (CodecNameMatches(name, external_codecs[c].name)) {
return true;
}
}
return false;
}
std::vector<WebRtcVideoChannel2::VideoCodecSettings>
WebRtcVideoChannel2::FilterSupportedCodecs(
const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
const {
std::vector<VideoCodecSettings> supported_codecs;
for (size_t i = 0; i < mapped_codecs.size(); ++i) {
const VideoCodecSettings& codec = mapped_codecs[i];
if (CodecIsInternallySupported(codec.codec.name) ||
CodecIsExternallySupported(codec.codec.name)) {
supported_codecs.push_back(codec);
}
}
return supported_codecs;
}
bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvCodecs");
LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
if (!ValidateCodecFormats(codecs)) {
return false;
}
const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs);
if (mapped_codecs.empty()) {
LOG(LS_ERROR) << "SetRecvCodecs called without any video codecs.";
return false;
}
const std::vector<VideoCodecSettings> supported_codecs =
FilterSupportedCodecs(mapped_codecs);
if (mapped_codecs.size() != supported_codecs.size()) {
LOG(LS_ERROR) << "SetRecvCodecs called with unsupported video codecs.";
return false;
}
recv_codecs_ = supported_codecs;
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
it != receive_streams_.end();
++it) {
it->second->SetRecvCodecs(recv_codecs_);
}
return true;
}
bool WebRtcVideoChannel2::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendCodecs");
LOG(LS_INFO) << "SetSendCodecs: " << CodecVectorToString(codecs);
if (!ValidateCodecFormats(codecs)) {
return false;
}
const std::vector<VideoCodecSettings> supported_codecs =
FilterSupportedCodecs(MapCodecs(codecs));
if (supported_codecs.empty()) {
LOG(LS_ERROR) << "No video codecs supported by encoder factory.";
return false;
}
LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
VideoCodecSettings old_codec;
if (send_codec_.Get(&old_codec) && supported_codecs.front() == old_codec) {
// Using same codec, avoid reconfiguring.
return true;
}
send_codec_.Set(supported_codecs.front());
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
assert(it->second != NULL);
it->second->SetCodec(supported_codecs.front());
}
VideoCodec codec = supported_codecs.front().codec;
int bitrate_kbps;
if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
bitrate_config_.min_bitrate_bps = bitrate_kbps * 1000;
} else {
bitrate_config_.min_bitrate_bps = 0;
}
if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
bitrate_config_.start_bitrate_bps = bitrate_kbps * 1000;
} else {
// Do not reconfigure start bitrate unless it's specified and positive.
bitrate_config_.start_bitrate_bps = -1;
}
if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
bitrate_config_.max_bitrate_bps = bitrate_kbps * 1000;
} else {
bitrate_config_.max_bitrate_bps = -1;
}
call_->SetBitrateConfig(bitrate_config_);
return true;
}
bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
VideoCodecSettings codec_settings;
if (!send_codec_.Get(&codec_settings)) {
LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
return false;
}
*codec = codec_settings.codec;
return true;
}
bool WebRtcVideoChannel2::SetSendStreamFormat(uint32 ssrc,
const VideoFormat& format) {
LOG(LS_VERBOSE) << "SetSendStreamFormat:" << ssrc << " -> "
<< format.ToString();
rtc::CritScope stream_lock(&stream_crit_);
if (send_streams_.find(ssrc) == send_streams_.end()) {
return false;
}
return send_streams_[ssrc]->SetVideoFormat(format);
}
bool WebRtcVideoChannel2::SetRender(bool render) {
// TODO(pbos): Implement. Or refactor away as it shouldn't be needed.
LOG(LS_VERBOSE) << "SetRender: " << (render ? "true" : "false");
return true;
}
bool WebRtcVideoChannel2::SetSend(bool send) {
LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
if (send && !send_codec_.IsSet()) {
LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
return false;
}
if (send) {
StartAllSendStreams();
} else {
StopAllSendStreams();
}
sending_ = send;
return true;
}
bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters.";
return false;
}
uint32 ssrc = sp.first_ssrc();
assert(ssrc != 0);
// TODO(pbos): Make sure none of sp.ssrcs are used, not just the identifying
// ssrc.
rtc::CritScope stream_lock(&stream_crit_);
if (send_streams_.find(ssrc) != send_streams_.end()) {
LOG(LS_ERROR) << "Send stream with ssrc '" << ssrc << "' already exists.";
return false;
}
std::vector<uint32> primary_ssrcs;
sp.GetPrimarySsrcs(&primary_ssrcs);
std::vector<uint32> rtx_ssrcs;
sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs);
if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
LOG(LS_ERROR)
<< "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
<< sp.ToString();
return false;
}
WebRtcVideoSendStream* stream =
new WebRtcVideoSendStream(call_.get(),
external_encoder_factory_,
options_,
send_codec_,
sp,
send_rtp_extensions_);
send_streams_[ssrc] = stream;
if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
rtcp_receiver_report_ssrc_ = ssrc;
}
if (default_send_ssrc_ == 0) {
default_send_ssrc_ = ssrc;
}
if (sending_) {
stream->Start();
}
return true;
}
bool WebRtcVideoChannel2::RemoveSendStream(uint32 ssrc) {
LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
if (ssrc == 0) {
if (default_send_ssrc_ == 0) {
LOG(LS_ERROR) << "No default send stream active.";
return false;
}
LOG(LS_VERBOSE) << "Removing default stream: " << default_send_ssrc_;
ssrc = default_send_ssrc_;
}
WebRtcVideoSendStream* removed_stream;
{
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.find(ssrc);
if (it == send_streams_.end()) {
return false;
}
removed_stream = it->second;
send_streams_.erase(it);
}
delete removed_stream;
if (ssrc == default_send_ssrc_) {
default_send_ssrc_ = 0;
}
return true;
}
bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) {
LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
assert(sp.ssrcs.size() > 0);
uint32 ssrc = sp.first_ssrc();
assert(ssrc != 0); // TODO(pbos): Is this ever valid?
// TODO(pbos): Check if any of the SSRCs overlap.
rtc::CritScope stream_lock(&stream_crit_);
if (receive_streams_.find(ssrc) != receive_streams_.end()) {
LOG(LS_ERROR) << "Receive stream for SSRC " << ssrc << "already exists.";
return false;
}
webrtc::VideoReceiveStream::Config config;
ConfigureReceiverRtp(&config, sp);
// Set up A/V sync if there is a VoiceChannel.
// TODO(pbos): The A/V is synched by the receiving channel. So we need to know
// the SSRC of the remote audio channel in order to sync the correct webrtc
// VoiceEngine channel. For now sync the first channel in non-conference to
// match existing behavior in WebRtcVideoEngine.
if (voice_channel_ != NULL && receive_streams_.empty() &&
!options_.conference_mode.GetWithDefaultIfUnset(false)) {
config.audio_channel_id =
static_cast<WebRtcVoiceMediaChannel*>(voice_channel_)->voe_channel();
}
receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
call_.get(), external_decoder_factory_, config, recv_codecs_);
return true;
}
void WebRtcVideoChannel2::ConfigureReceiverRtp(
webrtc::VideoReceiveStream::Config* config,
const StreamParams& sp) const {
uint32 ssrc = sp.first_ssrc();
config->rtp.remote_ssrc = ssrc;
config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
config->rtp.extensions = recv_rtp_extensions_;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
// corresponding sender SSRC. Figure out what to do when we don't have
// (receive-only) or know a good local SSRC.
if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
} else {
config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
}
}
for (size_t i = 0; i < recv_codecs_.size(); ++i) {
MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
}
for (size_t i = 0; i < recv_codecs_.size(); ++i) {
uint32 rtx_ssrc;
if (recv_codecs_[i].rtx_payload_type != -1 &&
sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
config->rtp.rtx[recv_codecs_[i].codec.id];
rtx.ssrc = rtx_ssrc;
rtx.payload_type = recv_codecs_[i].rtx_payload_type;
}
}
}
bool WebRtcVideoChannel2::RemoveRecvStream(uint32 ssrc) {
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
if (ssrc == 0) {
LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
return false;
}
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32, WebRtcVideoReceiveStream*>::iterator stream =
receive_streams_.find(ssrc);
if (stream == receive_streams_.end()) {
LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
return false;
}
delete stream->second;
receive_streams_.erase(stream);
return true;
}
bool WebRtcVideoChannel2::SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
LOG(LS_INFO) << "SetRenderer: ssrc:" << ssrc << " "
<< (renderer ? "(ptr)" : "NULL");
if (ssrc == 0) {
default_unsignalled_ssrc_handler_.SetDefaultRenderer(this, renderer);
return true;
}
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
return false;
}
it->second->SetRenderer(renderer);
return true;
}
bool WebRtcVideoChannel2::GetRenderer(uint32 ssrc, VideoRenderer** renderer) {
if (ssrc == 0) {
*renderer = default_unsignalled_ssrc_handler_.GetDefaultRenderer();
return *renderer != NULL;
}
rtc::CritScope stream_lock(&stream_crit_);
std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.find(ssrc);
if (it == receive_streams_.end()) {
return false;
}
*renderer = it->second->GetRenderer();
return true;
}
bool WebRtcVideoChannel2::GetStats(const StatsOptions& options,
VideoMediaInfo* info) {
info->Clear();
FillSenderStats(info);
FillReceiverStats(info);
webrtc::Call::Stats stats = call_->GetStats();
FillBandwidthEstimationStats(stats, info);
if (stats.rtt_ms != -1) {
for (size_t i = 0; i < info->senders.size(); ++i) {
info->senders[i].rtt_ms = stats.rtt_ms;
}
}
return true;
}
void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
video_media_info->senders.push_back(it->second->GetVideoSenderInfo());
}
}
void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
it != receive_streams_.end();
++it) {
video_media_info->receivers.push_back(it->second->GetVideoReceiverInfo());
}
}
void WebRtcVideoChannel2::FillBandwidthEstimationStats(
const webrtc::Call::Stats& stats,
VideoMediaInfo* video_media_info) {
BandwidthEstimationInfo bwe_info;
bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
bwe_info.bucket_delay = stats.pacer_delay_ms;
// Get send stream bitrate stats.
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator stream =
send_streams_.begin();
stream != send_streams_.end();
++stream) {
stream->second->FillBandwidthEstimationInfo(&bwe_info);
}
video_media_info->bw_estimations.push_back(bwe_info);
}
bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
<< (capturer != NULL ? "(capturer)" : "NULL");
assert(ssrc != 0);
rtc::CritScope stream_lock(&stream_crit_);
if (send_streams_.find(ssrc) == send_streams_.end()) {
LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
return false;
}
return send_streams_[ssrc]->SetCapturer(capturer);
}
bool WebRtcVideoChannel2::SendIntraFrame() {
// TODO(pbos): Implement.
LOG(LS_VERBOSE) << "SendIntraFrame().";
return true;
}
bool WebRtcVideoChannel2::RequestIntraFrame() {
// TODO(pbos): Implement.
LOG(LS_VERBOSE) << "SendIntraFrame().";
return true;
}
void WebRtcVideoChannel2::OnPacketReceived(
rtc::Buffer* packet,
const rtc::PacketTime& packet_time) {
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(
reinterpret_cast<const uint8_t*>(packet->data()), packet->length());
switch (delivery_result) {
case webrtc::PacketReceiver::DELIVERY_OK:
return;
case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
return;
case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
break;
}
uint32 ssrc = 0;
if (!GetRtpSsrc(packet->data(), packet->length(), &ssrc)) {
return;
}
// TODO(pbos): Make sure that the unsignalled SSRC uses the video payload.
// Also figure out whether RTX needs to be handled.
switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
case UnsignalledSsrcHandler::kDropPacket:
return;
case UnsignalledSsrcHandler::kDeliverPacket:
break;
}
if (call_->Receiver()->DeliverPacket(
reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) !=
webrtc::PacketReceiver::DELIVERY_OK) {
LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
return;
}
}
void WebRtcVideoChannel2::OnRtcpReceived(
rtc::Buffer* packet,
const rtc::PacketTime& packet_time) {
if (call_->Receiver()->DeliverPacket(
reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) !=
webrtc::PacketReceiver::DELIVERY_OK) {
LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
}
}
void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
call_->SignalNetworkState(ready ? webrtc::Call::kNetworkUp
: webrtc::Call::kNetworkDown);
}
bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
<< (mute ? "mute" : "unmute");
assert(ssrc != 0);
rtc::CritScope stream_lock(&stream_crit_);
if (send_streams_.find(ssrc) == send_streams_.end()) {
LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
return false;
}
send_streams_[ssrc]->MuteStream(mute);
return true;
}
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
<< RtpExtensionsToString(extensions);
if (!ValidateRtpHeaderExtensionIds(extensions))
return false;
std::vector<webrtc::RtpExtension> filtered_extensions =
FilterRtpExtensions(extensions);
if (!RtpExtensionsHaveChanged(recv_rtp_extensions_, filtered_extensions))
return true;
recv_rtp_extensions_ = filtered_extensions;
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
it != receive_streams_.end();
++it) {
it->second->SetRtpExtensions(recv_rtp_extensions_);
}
return true;
}
bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
<< RtpExtensionsToString(extensions);
if (!ValidateRtpHeaderExtensionIds(extensions))
return false;
std::vector<webrtc::RtpExtension> filtered_extensions =
FilterRtpExtensions(extensions);
if (!RtpExtensionsHaveChanged(send_rtp_extensions_, filtered_extensions))
return true;
send_rtp_extensions_ = filtered_extensions;
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
it->second->SetRtpExtensions(send_rtp_extensions_);
}
return true;
}
bool WebRtcVideoChannel2::SetMaxSendBandwidth(int max_bitrate_bps) {
LOG(LS_INFO) << "SetMaxSendBandwidth: " << max_bitrate_bps << "bps.";
if (max_bitrate_bps <= 0) {
// Unsetting max bitrate.
max_bitrate_bps = -1;
}
bitrate_config_.start_bitrate_bps = -1;
bitrate_config_.max_bitrate_bps = max_bitrate_bps;
if (max_bitrate_bps > 0 &&
bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
bitrate_config_.min_bitrate_bps = max_bitrate_bps;
}
call_->SetBitrateConfig(bitrate_config_);
return true;
}
bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
LOG(LS_INFO) << "SetOptions: " << options.ToString();
VideoOptions old_options = options_;
options_.SetAll(options);
if (options_ == old_options) {
// No new options to set.
return true;
}
rtc::DiffServCodePoint dscp = options_.dscp.GetWithDefaultIfUnset(false)
? rtc::DSCP_AF41
: rtc::DSCP_DEFAULT;
MediaChannel::SetDscp(dscp);
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
it->second->SetOptions(options_);
}
return true;
}
void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
MediaChannel::SetInterface(iface);
// Set the RTP recv/send buffer to a bigger size
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
// Speculative change to increase the outbound socket buffer size.
// In b/15152257, we are seeing a significant number of packets discarded
// due to lack of socket buffer space, although it's not yet clear what the
// ideal value should be.
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_SNDBUF,
kVideoRtpBufferSize);
}
void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) {
// TODO(pbos): Implement.
}
void WebRtcVideoChannel2::OnMessage(rtc::Message* msg) {
// Ignored.
}
void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
it->second->OnCpuResolutionRequest(load == kOveruse
? CoordinatedVideoAdapter::DOWNGRADE
: CoordinatedVideoAdapter::UPGRADE);
}
}
bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, size_t len) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendPacket(&packet);
}
bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendRtcp(&packet);
}
void WebRtcVideoChannel2::StartAllSendStreams() {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
it->second->Start();
}
}
void WebRtcVideoChannel2::StopAllSendStreams() {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
it != send_streams_.end();
++it) {
it->second->Stop();
}
}
WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
VideoSendStreamParameters(
const webrtc::VideoSendStream::Config& config,
const VideoOptions& options,
const Settable<VideoCodecSettings>& codec_settings)
: config(config), options(options), codec_settings(codec_settings) {
}
WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
webrtc::Call* call,
WebRtcVideoEncoderFactory* external_encoder_factory,
const VideoOptions& options,
const Settable<VideoCodecSettings>& codec_settings,
const StreamParams& sp,
const std::vector<webrtc::RtpExtension>& rtp_extensions)
: call_(call),
external_encoder_factory_(external_encoder_factory),
stream_(NULL),
parameters_(webrtc::VideoSendStream::Config(), options, codec_settings),
allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
capturer_(NULL),
sending_(false),
muted_(false) {
parameters_.config.rtp.max_packet_size = kVideoMtu;
sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
&parameters_.config.rtp.rtx.ssrcs);
parameters_.config.rtp.c_name = sp.cname;
parameters_.config.rtp.extensions = rtp_extensions;
VideoCodecSettings params;
if (codec_settings.Get(&params)) {
SetCodec(params);
}
}
WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
DisconnectCapturer();
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
}
DestroyVideoEncoder(&allocated_encoder_);
}
static void SetWebRtcFrameToBlack(webrtc::I420VideoFrame* video_frame) {
assert(video_frame != NULL);
memset(video_frame->buffer(webrtc::kYPlane),
16,
video_frame->allocated_size(webrtc::kYPlane));
memset(video_frame->buffer(webrtc::kUPlane),
128,
video_frame->allocated_size(webrtc::kUPlane));
memset(video_frame->buffer(webrtc::kVPlane),
128,
video_frame->allocated_size(webrtc::kVPlane));
}
static void CreateBlackFrame(webrtc::I420VideoFrame* video_frame,
int width,
int height) {
video_frame->CreateEmptyFrame(
width, height, width, (width + 1) / 2, (width + 1) / 2);
SetWebRtcFrameToBlack(video_frame);
}
static void ConvertToI420VideoFrame(const VideoFrame& frame,
webrtc::I420VideoFrame* i420_frame) {
i420_frame->CreateFrame(
static_cast<int>(frame.GetYPitch() * frame.GetHeight()),
frame.GetYPlane(),
static_cast<int>(frame.GetUPitch() * ((frame.GetHeight() + 1) / 2)),
frame.GetUPlane(),
static_cast<int>(frame.GetVPitch() * ((frame.GetHeight() + 1) / 2)),
frame.GetVPlane(),
static_cast<int>(frame.GetWidth()),
static_cast<int>(frame.GetHeight()),
static_cast<int>(frame.GetYPitch()),
static_cast<int>(frame.GetUPitch()),
static_cast<int>(frame.GetVPitch()));
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
VideoCapturer* capturer,
const VideoFrame* frame) {
LOG(LS_VERBOSE) << "InputFrame: " << frame->GetWidth() << "x"
<< frame->GetHeight();
// Lock before copying, can be called concurrently when swapping input source.
rtc::CritScope frame_cs(&frame_lock_);
ConvertToI420VideoFrame(*frame, &video_frame_);
rtc::CritScope cs(&lock_);
if (stream_ == NULL) {
LOG(LS_WARNING) << "Capturer inputting frames before send codecs are "
"configured, dropping.";
return;
}
if (format_.width == 0) { // Dropping frames.
assert(format_.height == 0);
LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
return;
}
if (muted_) {
// Create a black frame to transmit instead.
CreateBlackFrame(&video_frame_,
static_cast<int>(frame->GetWidth()),
static_cast<int>(frame->GetHeight()));
}
// Reconfigure codec if necessary.
SetDimensions(
video_frame_.width(), video_frame_.height(), capturer->IsScreencast());
LOG(LS_VERBOSE) << "SwapFrame: " << video_frame_.width() << "x"
<< video_frame_.height() << " -> (codec) "
<< parameters_.encoder_config.streams.back().width << "x"
<< parameters_.encoder_config.streams.back().height;
stream_->Input()->SwapFrame(&video_frame_);
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
VideoCapturer* capturer) {
if (!DisconnectCapturer() && capturer == NULL) {
return false;
}
{
rtc::CritScope cs(&lock_);
if (capturer == NULL) {
if (stream_ != NULL) {
LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
webrtc::I420VideoFrame black_frame;
// TODO(pbos): Base width/height on last_dimensions_. This will however
// fail the test AddRemoveCapturer which needs to be fixed to permit
// sending black frames in the same size that was previously sent.
int width = format_.width;
int height = format_.height;
int half_width = (width + 1) / 2;
black_frame.CreateEmptyFrame(
width, height, width, half_width, half_width);
SetWebRtcFrameToBlack(&black_frame);
SetDimensions(width, height, last_dimensions_.is_screencast);
stream_->Input()->SwapFrame(&black_frame);
}
capturer_ = NULL;
return true;
}
capturer_ = capturer;
}
// Lock cannot be held while connecting the capturer to prevent lock-order
// violations.
capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame);
return true;
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat(
const VideoFormat& format) {
if ((format.width == 0 || format.height == 0) &&
format.width != format.height) {
LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not "
"both, 0x0 drops frames).";
return false;
}
rtc::CritScope cs(&lock_);
if (format.width == 0 && format.height == 0) {
LOG(LS_INFO)
<< "0x0 resolution selected. Captured frames will be dropped for ssrc: "
<< parameters_.config.rtp.ssrcs[0] << ".";
} else {
// TODO(pbos): Fix me, this only affects the last stream!
parameters_.encoder_config.streams.back().max_framerate =
VideoFormat::IntervalToFps(format.interval);
SetDimensions(format.width, format.height, false);
}
format_ = format;
return true;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
rtc::CritScope cs(&lock_);
muted_ = mute;
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
cricket::VideoCapturer* capturer;
{
rtc::CritScope cs(&lock_);
if (capturer_ == NULL) {
return false;
}
capturer = capturer_;
capturer_ = NULL;
}
capturer->SignalVideoFrame.disconnect(this);
return true;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
const VideoOptions& options) {
rtc::CritScope cs(&lock_);
VideoCodecSettings codec_settings;
if (parameters_.codec_settings.Get(&codec_settings)) {
SetCodecAndOptions(codec_settings, options);
} else {
parameters_.options = options;
}
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
const VideoCodecSettings& codec_settings) {
rtc::CritScope cs(&lock_);
SetCodecAndOptions(codec_settings, parameters_.options);
}
webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
if (CodecNameMatches(name, kVp8CodecName)) {
return webrtc::kVideoCodecVP8;
} else if (CodecNameMatches(name, kVp9CodecName)) {
return webrtc::kVideoCodecVP9;
} else if (CodecNameMatches(name, kH264CodecName)) {
return webrtc::kVideoCodecH264;
}
return webrtc::kVideoCodecUnknown;
}
WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
const VideoCodec& codec) {
webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
// Do not re-create encoders of the same type.
if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) {
return allocated_encoder_;
}
if (external_encoder_factory_ != NULL) {
webrtc::VideoEncoder* encoder =
external_encoder_factory_->CreateVideoEncoder(type);
if (encoder != NULL) {
return AllocatedEncoder(encoder, type, true);
}
}
if (type == webrtc::kVideoCodecVP8) {
return AllocatedEncoder(
webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false);
} else if (type == webrtc::kVideoCodecVP9) {
return AllocatedEncoder(
webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
}
// This shouldn't happen, we should not be trying to create something we don't
// support.
assert(false);
return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
AllocatedEncoder* encoder) {
if (encoder->external) {
external_encoder_factory_->DestroyVideoEncoder(encoder->encoder);
} else {
delete encoder->encoder;
}
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
const VideoCodecSettings& codec_settings,
const VideoOptions& options) {
if (last_dimensions_.width == -1) {
last_dimensions_.width = codec_settings.codec.width;
last_dimensions_.height = codec_settings.codec.height;
last_dimensions_.is_screencast = false;
}
parameters_.encoder_config =
CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
if (parameters_.encoder_config.streams.empty()) {
return;
}
format_ = VideoFormat(codec_settings.codec.width,
codec_settings.codec.height,
VideoFormat::FpsToInterval(30),
FOURCC_I420);
AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
parameters_.config.encoder_settings.encoder = new_encoder.encoder;
parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
parameters_.config.rtp.fec = codec_settings.fec;
// Set RTX payload type if RTX is enabled.
if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
}
if (IsNackEnabled(codec_settings.codec)) {
parameters_.config.rtp.nack.rtp_history_ms = kNackHistoryMs;
}
options.suspend_below_min_bitrate.Get(
&parameters_.config.suspend_below_min_bitrate);
parameters_.codec_settings.Set(codec_settings);
parameters_.options = options;
RecreateWebRtcStream();
if (allocated_encoder_.encoder != new_encoder.encoder) {
DestroyVideoEncoder(&allocated_encoder_);
allocated_encoder_ = new_encoder;
}
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions(
const std::vector<webrtc::RtpExtension>& rtp_extensions) {
rtc::CritScope cs(&lock_);
parameters_.config.rtp.extensions = rtp_extensions;
RecreateWebRtcStream();
}
webrtc::VideoEncoderConfig
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
const Dimensions& dimensions,
const VideoCodec& codec) const {
webrtc::VideoEncoderConfig encoder_config;
if (dimensions.is_screencast) {
int screencast_min_bitrate_kbps;
parameters_.options.screencast_min_bitrate.Get(
&screencast_min_bitrate_kbps);
encoder_config.min_transmit_bitrate_bps =
screencast_min_bitrate_kbps * 1000;
encoder_config.content_type = webrtc::VideoEncoderConfig::kScreenshare;
} else {
encoder_config.min_transmit_bitrate_bps = 0;
encoder_config.content_type = webrtc::VideoEncoderConfig::kRealtimeVideo;
}
// Restrict dimensions according to codec max.
int width = dimensions.width;
int height = dimensions.height;
if (!dimensions.is_screencast) {
if (codec.width < width)
width = codec.width;
if (codec.height < height)
height = codec.height;
}
VideoCodec clamped_codec = codec;
clamped_codec.width = width;
clamped_codec.height = height;
encoder_config.streams = CreateVideoStreams(
clamped_codec, parameters_.options, parameters_.config.rtp.ssrcs.size());
// Conference mode screencast uses 2 temporal layers split at 100kbit.
if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) &&
dimensions.is_screencast && encoder_config.streams.size() == 1) {
ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
// For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
// on the VideoCodec struct as target and max bitrates, respectively.
// See eg. webrtc::VP8EncoderImpl::SetRates().
encoder_config.streams[0].target_bitrate_bps =
config.tl0_bitrate_kbps * 1000;
encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
config.tl0_bitrate_kbps * 1000);
}
return encoder_config;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
int width,
int height,
bool is_screencast) {
if (last_dimensions_.width == width && last_dimensions_.height == height &&
last_dimensions_.is_screencast == is_screencast) {
// Configured using the same parameters, do not reconfigure.
return;
}
LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
<< (is_screencast ? " (screencast)" : " (not screencast)");
last_dimensions_.width = width;
last_dimensions_.height = height;
last_dimensions_.is_screencast = is_screencast;
assert(!parameters_.encoder_config.streams.empty());
VideoCodecSettings codec_settings;
parameters_.codec_settings.Get(&codec_settings);
webrtc::VideoEncoderConfig encoder_config =
CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
encoder_config.encoder_specific_settings =
ConfigureVideoEncoderSettings(codec_settings.codec, parameters_.options);
bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
encoder_config.encoder_specific_settings = NULL;
if (!stream_reconfigured) {
LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
<< width << "x" << height;
return;
}
parameters_.encoder_config = encoder_config;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
rtc::CritScope cs(&lock_);
assert(stream_ != NULL);
stream_->Start();
sending_ = true;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
rtc::CritScope cs(&lock_);
if (stream_ != NULL) {
stream_->Stop();
}
sending_ = false;
}
VideoSenderInfo
WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
VideoSenderInfo info;
rtc::CritScope cs(&lock_);
for (size_t i = 0; i < parameters_.config.rtp.ssrcs.size(); ++i) {
info.add_ssrc(parameters_.config.rtp.ssrcs[i]);
}
if (stream_ == NULL) {
return info;
}
webrtc::VideoSendStream::Stats stats = stream_->GetStats();
info.framerate_input = stats.input_frame_rate;
info.framerate_sent = stats.encode_frame_rate;
info.send_frame_width = 0;
info.send_frame_height = 0;
for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end();
++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
webrtc::SsrcStats stream_stats = it->second;
info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
if (stream_stats.sent_width > info.send_frame_width)
info.send_frame_width = stream_stats.sent_width;
if (stream_stats.sent_height > info.send_frame_height)
info.send_frame_height = stream_stats.sent_height;
}
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
webrtc::SsrcStats first_stream_stats = stats.substreams.begin()->second;
info.fraction_lost =
static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
(1 << 8);
}
if (capturer_ != NULL && !capturer_->IsMuted()) {
VideoFormat last_captured_frame_format;
capturer_->GetStats(&info.adapt_frame_drops,
&info.effects_frame_drops,
&info.capturer_frame_time,
&last_captured_frame_format);
info.input_frame_width = last_captured_frame_format.width;
info.input_frame_height = last_captured_frame_format.height;
}
// TODO(pbos): Support or remove the following stats.
info.packets_cached = -1;
return info;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo(
BandwidthEstimationInfo* bwe_info) {
rtc::CritScope cs(&lock_);
if (stream_ == NULL) {
return;
}
webrtc::VideoSendStream::Stats stats = stream_->GetStats();
for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end();
++it) {
bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
}
bwe_info->actual_enc_bitrate = stats.media_bitrate_bps;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::OnCpuResolutionRequest(
CoordinatedVideoAdapter::AdaptRequest adapt_request) {
rtc::CritScope cs(&lock_);
bool adapt_cpu;
parameters_.options.cpu_overuse_detection.Get(&adapt_cpu);
if (!adapt_cpu) {
return;
}
if (capturer_ == NULL || capturer_->video_adapter() == NULL) {
return;
}
capturer_->video_adapter()->OnCpuResolutionRequest(adapt_request);
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoSendStream(stream_);
}
VideoCodecSettings codec_settings;
parameters_.codec_settings.Get(&codec_settings);
parameters_.encoder_config.encoder_specific_settings =
ConfigureVideoEncoderSettings(codec_settings.codec, parameters_.options);
stream_ = call_->CreateVideoSendStream(parameters_.config,
parameters_.encoder_config);
parameters_.encoder_config.encoder_specific_settings = NULL;
if (sending_) {
stream_->Start();
}
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
webrtc::Call* call,
WebRtcVideoDecoderFactory* external_decoder_factory,
const webrtc::VideoReceiveStream::Config& config,
const std::vector<VideoCodecSettings>& recv_codecs)
: call_(call),
stream_(NULL),
config_(config),
external_decoder_factory_(external_decoder_factory),
renderer_(NULL),
last_width_(-1),
last_height_(-1),
first_frame_timestamp_(-1),
estimated_remote_start_ntp_time_ms_(0) {
config_.renderer = this;
// SetRecvCodecs will also reset (start) the VideoReceiveStream.
SetRecvCodecs(recv_codecs);
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
call_->DestroyVideoReceiveStream(stream_);
ClearDecoders(&allocated_decoders_);
}
WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
std::vector<AllocatedDecoder>* old_decoders,
const VideoCodec& codec) {
webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
for (size_t i = 0; i < old_decoders->size(); ++i) {
if ((*old_decoders)[i].type == type) {
AllocatedDecoder decoder = (*old_decoders)[i];
(*old_decoders)[i] = old_decoders->back();
old_decoders->pop_back();
return decoder;
}
}
if (external_decoder_factory_ != NULL) {
webrtc::VideoDecoder* decoder =
external_decoder_factory_->CreateVideoDecoder(type);
if (decoder != NULL) {
return AllocatedDecoder(decoder, type, true);
}
}
if (type == webrtc::kVideoCodecVP8) {
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
}
// This shouldn't happen, we should not be trying to create something we don't
// support.
assert(false);
return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false);
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs(
const std::vector<VideoCodecSettings>& recv_codecs) {
std::vector<AllocatedDecoder> old_decoders = allocated_decoders_;
allocated_decoders_.clear();
config_.decoders.clear();
for (size_t i = 0; i < recv_codecs.size(); ++i) {
AllocatedDecoder allocated_decoder =
CreateOrReuseVideoDecoder(&old_decoders, recv_codecs[i].codec);
allocated_decoders_.push_back(allocated_decoder);
webrtc::VideoReceiveStream::Decoder decoder;
decoder.decoder = allocated_decoder.decoder;
decoder.payload_type = recv_codecs[i].codec.id;
decoder.payload_name = recv_codecs[i].codec.name;
config_.decoders.push_back(decoder);
}
// TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
config_.rtp.fec = recv_codecs.front().fec;
config_.rtp.nack.rtp_history_ms =
IsNackEnabled(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
config_.rtp.remb = IsRembEnabled(recv_codecs.begin()->codec);
ClearDecoders(&old_decoders);
RecreateWebRtcStream();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions) {
config_.rtp.extensions = extensions;
RecreateWebRtcStream();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoReceiveStream(stream_);
}
stream_ = call_->CreateVideoReceiveStream(config_);
stream_->Start();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
std::vector<AllocatedDecoder>* allocated_decoders) {
for (size_t i = 0; i < allocated_decoders->size(); ++i) {
if ((*allocated_decoders)[i].external) {
external_decoder_factory_->DestroyVideoDecoder(
(*allocated_decoders)[i].decoder);
} else {
delete (*allocated_decoders)[i].decoder;
}
}
allocated_decoders->clear();
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame(
const webrtc::I420VideoFrame& frame,
int time_to_render_ms) {
rtc::CritScope crit(&renderer_lock_);
if (first_frame_timestamp_ < 0)
first_frame_timestamp_ = frame.timestamp();
int64_t rtp_time_elapsed_since_first_frame =
(timestamp_wraparound_handler_.Unwrap(frame.timestamp()) -
first_frame_timestamp_);
int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame /
(cricket::kVideoCodecClockrate / 1000);
if (frame.ntp_time_ms() > 0)
estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
if (renderer_ == NULL) {
LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoRenderer.";
return;
}
if (frame.width() != last_width_ || frame.height() != last_height_) {
SetSize(frame.width(), frame.height());
}
LOG(LS_VERBOSE) << "RenderFrame: (" << frame.width() << "x" << frame.height()
<< ")";
const WebRtcVideoRenderFrame render_frame(&frame, elapsed_time_ms);
renderer_->RenderFrame(&render_frame);
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRenderer(
cricket::VideoRenderer* renderer) {
rtc::CritScope crit(&renderer_lock_);
renderer_ = renderer;
if (renderer_ != NULL && last_width_ != -1) {
SetSize(last_width_, last_height_);
}
}
VideoRenderer* WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetRenderer() {
// TODO(pbos): Remove GetRenderer and all uses of it, it's thread-unsafe by
// design.
rtc::CritScope crit(&renderer_lock_);
return renderer_;
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSize(int width,
int height) {
rtc::CritScope crit(&renderer_lock_);
if (!renderer_->SetSize(width, height, 0)) {
LOG(LS_ERROR) << "Could not set renderer size.";
}
last_width_ = width;
last_height_ = height;
}
VideoReceiverInfo
WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
VideoReceiverInfo info;
info.add_ssrc(config_.rtp.remote_ssrc);
webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
stats.rtp_stats.transmitted.header_bytes +
stats.rtp_stats.transmitted.padding_bytes;
info.packets_rcvd = stats.rtp_stats.transmitted.packets;
info.framerate_rcvd = stats.network_frame_rate;
info.framerate_decoded = stats.decode_frame_rate;
info.framerate_output = stats.render_frame_rate;
rtc::CritScope frame_cs(&renderer_lock_);
info.frame_width = last_width_;
info.frame_height = last_height_;
info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
// TODO(pbos): Support or remove the following stats.
info.packets_concealed = -1;
return info;
}
WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
: rtx_payload_type(-1) {}
bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
const WebRtcVideoChannel2::VideoCodecSettings& other) const {
return codec == other.codec &&
fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
fec.red_payload_type == other.fec.red_payload_type &&
rtx_payload_type == other.rtx_payload_type;
}
std::vector<WebRtcVideoChannel2::VideoCodecSettings>
WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
assert(!codecs.empty());
std::vector<VideoCodecSettings> video_codecs;
std::map<int, bool> payload_used;
std::map<int, VideoCodec::CodecType> payload_codec_type;
std::map<int, int> rtx_mapping; // video payload type -> rtx payload type.
webrtc::FecConfig fec_settings;
for (size_t i = 0; i < codecs.size(); ++i) {
const VideoCodec& in_codec = codecs[i];
int payload_type = in_codec.id;
if (payload_used[payload_type]) {
LOG(LS_ERROR) << "Payload type already registered: "
<< in_codec.ToString();
return std::vector<VideoCodecSettings>();
}
payload_used[payload_type] = true;
payload_codec_type[payload_type] = in_codec.GetCodecType();
switch (in_codec.GetCodecType()) {
case VideoCodec::CODEC_RED: {
// RED payload type, should not have duplicates.
assert(fec_settings.red_payload_type == -1);
fec_settings.red_payload_type = in_codec.id;
continue;
}
case VideoCodec::CODEC_ULPFEC: {
// ULPFEC payload type, should not have duplicates.
assert(fec_settings.ulpfec_payload_type == -1);
fec_settings.ulpfec_payload_type = in_codec.id;
continue;
}
case VideoCodec::CODEC_RTX: {
int associated_payload_type;
if (!in_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type)) {
LOG(LS_ERROR) << "RTX codec without associated payload type: "
<< in_codec.ToString();
return std::vector<VideoCodecSettings>();
}
rtx_mapping[associated_payload_type] = in_codec.id;
continue;
}
case VideoCodec::CODEC_VIDEO:
break;
}
video_codecs.push_back(VideoCodecSettings());
video_codecs.back().codec = in_codec;
}
// One of these codecs should have been a video codec. Only having FEC
// parameters into this code is a logic error.
assert(!video_codecs.empty());
for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
it != rtx_mapping.end();
++it) {
if (!payload_used[it->first]) {
LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
return std::vector<VideoCodecSettings>();
}
if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO) {
LOG(LS_ERROR) << "RTX not mapped to regular video codec.";
return std::vector<VideoCodecSettings>();
}
}
// TODO(pbos): Write tests that figure out that I have not verified that RTX
// codecs aren't mapped to bogus payloads.
for (size_t i = 0; i < video_codecs.size(); ++i) {
video_codecs[i].fec = fec_settings;
if (rtx_mapping[video_codecs[i].codec.id] != 0) {
video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
}
}
return video_codecs;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VIDEO
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