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a5f6fb53ba802fcf44e3c187d798c5a53ad555df
platform-external-webrtc/talk/media/webrtc/webrtcvideoengine2.cc
pbos@webrtc.org a5f6fb53ba Permit single-stream max bitrates above 2000k. 
BUG=4463
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49509004

Cr-Commit-Position: refs/heads/master@{#8839}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8839 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 22:30:11 +00:00

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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;
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;
int max_bitrate_kbps;
if (!codec.GetParam(kCodecParamMaxBitrate, &max_bitrate_kbps) ||
max_bitrate_kbps < kMaxVideoBitrate) {
max_bitrate_kbps = kMaxVideoBitrate;
}
stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_kbps * 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(
WebRtcVideoChannel2* 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, true)) {
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(WebRtcVoiceEngine* voice_engine)
: worker_thread_(NULL),
voice_engine_(voice_engine),
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].width = codec.width;
video_codecs_[i].height = codec.height;
video_codecs_[i].framerate = codec.framerate;
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 = std::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;
}
// 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;
}
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;
}
// External video encoders are given payloads 120-127. This also means that
// we only support up to 8 external payload types.
const int kExternalVideoPayloadTypeBase = 120;
size_t payload_type = kExternalVideoPayloadTypeBase + i;
assert(payload_type < 128);
VideoCodec codec(static_cast<int>(payload_type),
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_id_(voice_channel != nullptr
? static_cast<WebRtcVoiceMediaChannel*>(
voice_channel)->voe_channel()
: -1),
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) {
return AddRecvStream(sp, false);
}
bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
bool default_stream) {
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_);
{
auto it = receive_streams_.find(ssrc);
if (it != receive_streams_.end()) {
if (default_stream || !it->second->IsDefaultStream()) {
LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc
<< "' already exists.";
return false;
}
delete it->second;
receive_streams_.erase(it);
}
}
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_id_ != -1 && receive_streams_.empty() &&
!options_.conference_mode.GetWithDefaultIfUnset(false)) {
config.audio_channel_id = voice_channel_id_;
}
receive_streams_[ssrc] =
new WebRtcVideoReceiveStream(call_.get(), external_decoder_factory_,
default_stream, 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(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): Ignore unsignalled packets that don't use the video payload
// (prevent creating default receivers for RTX configured as if it would
// receive media payloads on those SSRCs).
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),
old_adapt_changes_(0) {
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 CreateBlackFrame(webrtc::I420VideoFrame* video_frame,
int width,
int height) {
video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
(width + 1) / 2);
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));
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
VideoCapturer* capturer,
const VideoFrame* frame) {
TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame");
LOG(LS_VERBOSE) << "InputFrame: " << frame->GetWidth() << "x"
<< frame->GetHeight();
webrtc::I420VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0,
frame->GetVideoRotation());
rtc::CritScope cs(&lock_);
if (stream_ == NULL) {
LOG(LS_WARNING) << "Capturer inputting frames before send codecs are "
"configured, dropping.";
return;
}
// Not sending, abort early to prevent expensive reconfigurations while
// setting up codecs etc.
if (!sending_)
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) << "IncomingCapturedFrame: " << video_frame.width() << "x"
<< video_frame.height() << " -> (codec) "
<< parameters_.encoder_config.streams.back().width << "x"
<< parameters_.encoder_config.streams.back().height;
stream_->Input()->IncomingCapturedFrame(video_frame);
}
bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
VideoCapturer* capturer) {
TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
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;
CreateBlackFrame(&black_frame, last_dimensions_.width,
last_dimensions_.height);
stream_->Input()->IncomingCapturedFrame(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;
if (capturer_->video_adapter() != nullptr)
old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes();
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) {
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()) {
if (codec_settings.rtx_payload_type == -1) {
LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
"payload type. Ignoring.";
parameters_.config.rtp.rtx.ssrcs.clear();
} else {
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;
webrtc::VideoSendStream::Stats stats;
{
rtc::CritScope cs(&lock_);
for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
info.add_ssrc(ssrc);
for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
if (i == parameters_.encoder_config.streams.size() - 1) {
info.preferred_bitrate +=
parameters_.encoder_config.streams[i].max_bitrate_bps;
} else {
info.preferred_bitrate +=
parameters_.encoder_config.streams[i].target_bitrate_bps;
}
}
if (stream_ == NULL)
return info;
stats = stream_->GetStats();
info.adapt_changes = old_adapt_changes_;
info.adapt_reason = CoordinatedVideoAdapter::ADAPTREASON_NONE;
if (capturer_ != NULL) {
if (!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;
}
if (capturer_->video_adapter() != nullptr) {
info.adapt_changes += capturer_->video_adapter()->adaptation_changes();
info.adapt_reason = capturer_->video_adapter()->adapt_reason();
}
}
}
info.framerate_input = stats.input_frame_rate;
info.framerate_sent = stats.encode_frame_rate;
info.avg_encode_ms = stats.avg_encode_time_ms;
info.encode_usage_percent = stats.encode_usage_percent;
info.nominal_bitrate = stats.media_bitrate_bps;
info.send_frame_width = 0;
info.send_frame_height = 0;
for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end(); ++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
webrtc::VideoSendStream::StreamStats 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.width > info.send_frame_width)
info.send_frame_width = stream_stats.width;
if (stream_stats.height > info.send_frame_height)
info.send_frame_height = stream_stats.height;
info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
}
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
webrtc::VideoSendStream::StreamStats first_stream_stats =
stats.substreams.begin()->second;
info.fraction_lost =
static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
(1 << 8);
}
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::VideoSendStream::StreamStats>::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->target_enc_bitrate += stats.target_media_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);
webrtc::VideoSendStream::Config config = parameters_.config;
if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
"payload type the set codec. Ignoring RTX.";
config.rtp.rtx.ssrcs.clear();
}
stream_ = call_->CreateVideoSendStream(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,
bool default_stream,
const webrtc::VideoReceiveStream::Config& config,
const std::vector<VideoCodecSettings>& recv_codecs)
: call_(call),
stream_(NULL),
default_stream_(default_stream),
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);
}
if (type == webrtc::kVideoCodecVP9) {
return AllocatedDecoder(
webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), 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 WebRtcVideoFrame render_frame(
frame.video_frame_buffer(),
elapsed_time_ms * rtc::kNumNanosecsPerMillisec,
frame.render_time_ms() * rtc::kNumNanosecsPerMillisec);
renderer_->RenderFrame(&render_frame);
}
bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsTextureSupported() const {
return true;
}
bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
return default_stream_;
}
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_;
}
info.decode_ms = stats.decode_ms;
info.max_decode_ms = stats.max_decode_ms;
info.current_delay_ms = stats.current_delay_ms;
info.target_delay_ms = stats.target_delay_ms;
info.jitter_buffer_ms = stats.jitter_buffer_ms;
info.min_playout_delay_ms = stats.min_playout_delay_ms;
info.render_delay_ms = stats.render_delay_ms;
info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
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;
// |rtx_mapping| maps video payload type to rtx payload type.
std::map<int, int> rtx_mapping;
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) ||
!IsValidRtpPayloadType(associated_payload_type)) {
LOG(LS_ERROR)
<< "RTX codec with invalid or no 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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