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webrtc / src / f5bebd40f38d3d35465dc6fc1f4c8f869688b048 / . / talk / media / webrtc / webrtcvideoengine.cc
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/*
* libjingle
* Copyright 2004 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/webrtcvideoengine.h"
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <math.h>
#include <set>
#include "talk/base/basictypes.h"
#include "talk/base/buffer.h"
#include "talk/base/byteorder.h"
#include "talk/base/common.h"
#include "talk/base/cpumonitor.h"
#include "talk/base/logging.h"
#include "talk/base/stringutils.h"
#include "talk/base/thread.h"
#include "talk/base/timeutils.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/videoadapter.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videorenderer.h"
#include "talk/media/devices/filevideocapturer.h"
#include "talk/media/webrtc/webrtcpassthroughrender.h"
#include "talk/media/webrtc/webrtctexturevideoframe.h"
#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvie.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "webrtc/experiments.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#if !defined(LIBPEERCONNECTION_LIB)
#ifndef HAVE_WEBRTC_VIDEO
#error Need webrtc video
#endif
#include "talk/media/webrtc/webrtcmediaengine.h"
WRME_EXPORT
cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
webrtc::AudioDeviceModule* adm, webrtc::AudioDeviceModule* adm_sc,
cricket::WebRtcVideoEncoderFactory* encoder_factory,
cricket::WebRtcVideoDecoderFactory* decoder_factory) {
return new cricket::WebRtcMediaEngine(adm, adm_sc, encoder_factory,
decoder_factory);
}
WRME_EXPORT
void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
delete static_cast<cricket::WebRtcMediaEngine*>(media_engine);
}
#endif
namespace cricket {
static const int kDefaultLogSeverity = talk_base::LS_WARNING;
static const int kMinVideoBitrate = 50;
static const int kStartVideoBitrate = 300;
static const int kMaxVideoBitrate = 2000;
static const int kDefaultConferenceModeMaxVideoBitrate = 500;
// Controlled by exp, try a super low minimum bitrate for poor connections.
static const int kLowerMinBitrate = 30;
static const int kVideoMtu = 1200;
static const int kVideoRtpBufferSize = 65536;
static const char kVp8PayloadName[] = "VP8";
static const char kRedPayloadName[] = "red";
static const char kFecPayloadName[] = "ulpfec";
static const int kDefaultNumberOfTemporalLayers = 1; // 1:1
static const int kMaxExternalVideoCodecs = 8;
static const int kExternalVideoPayloadTypeBase = 120;
// Static allocation of payload type values for external video codec.
static int GetExternalVideoPayloadType(int index) {
ASSERT(index >= 0 && index < kMaxExternalVideoCodecs);
return kExternalVideoPayloadTypeBase + index;
}
static void LogMultiline(talk_base::LoggingSeverity sev, char* text) {
const char* delim = "\r\n";
// TODO(fbarchard): Fix strtok lint warning.
for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
LOG_V(sev) << tok;
}
}
// Severity is an integer because it comes is assumed to be from command line.
static int SeverityToFilter(int severity) {
int filter = webrtc::kTraceNone;
switch (severity) {
case talk_base::LS_VERBOSE:
filter |= webrtc::kTraceAll;
case talk_base::LS_INFO:
filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
case talk_base::LS_WARNING:
filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
case talk_base::LS_ERROR:
filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
}
return filter;
}
static const int kCpuMonitorPeriodMs = 2000; // 2 seconds.
static const bool kNotSending = false;
// Default video dscp value.
// See http://tools.ietf.org/html/rfc2474 for details
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
static const talk_base::DiffServCodePoint kVideoDscpValue =
talk_base::DSCP_AF41;
static bool IsNackEnabled(const VideoCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
}
// Returns true if Receiver Estimated Max Bitrate is enabled.
static bool IsRembEnabled(const VideoCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamRemb,
kParamValueEmpty));
}
// TODO(mallinath) - Remove this after trunk of webrtc is pushed to GTP.
#if !defined(USE_WEBRTC_DEV_BRANCH)
bool operator==(const webrtc::VideoCodecVP8& lhs,
const webrtc::VideoCodecVP8& rhs) {
return lhs.pictureLossIndicationOn == rhs.pictureLossIndicationOn &&
lhs.feedbackModeOn == rhs.feedbackModeOn &&
lhs.complexity == rhs.complexity &&
lhs.resilience == rhs.resilience &&
lhs.numberOfTemporalLayers == rhs.numberOfTemporalLayers &&
lhs.denoisingOn == rhs.denoisingOn &&
lhs.errorConcealmentOn == rhs.errorConcealmentOn &&
lhs.automaticResizeOn == rhs.automaticResizeOn &&
lhs.frameDroppingOn == rhs.frameDroppingOn &&
lhs.keyFrameInterval == rhs.keyFrameInterval;
}
bool operator!=(const webrtc::VideoCodecVP8& lhs,
const webrtc::VideoCodecVP8& rhs) {
return !(lhs == rhs);
}
bool operator==(const webrtc::SimulcastStream& lhs,
const webrtc::SimulcastStream& rhs) {
return lhs.width == rhs.width &&
lhs.height == rhs.height &&
lhs.numberOfTemporalLayers == rhs.numberOfTemporalLayers &&
lhs.maxBitrate == rhs.maxBitrate &&
lhs.targetBitrate == rhs.targetBitrate &&
lhs.minBitrate == rhs.minBitrate &&
lhs.qpMax == rhs.qpMax;
}
bool operator!=(const webrtc::SimulcastStream& lhs,
const webrtc::SimulcastStream& rhs) {
return !(lhs == rhs);
}
bool operator==(const webrtc::VideoCodec& lhs,
const webrtc::VideoCodec& rhs) {
bool ret = lhs.codecType == rhs.codecType &&
(_stricmp(lhs.plName, rhs.plName) == 0) &&
lhs.plType == rhs.plType &&
lhs.width == rhs.width &&
lhs.height == rhs.height &&
lhs.startBitrate == rhs.startBitrate &&
lhs.maxBitrate == rhs.maxBitrate &&
lhs.minBitrate == rhs.minBitrate &&
lhs.maxFramerate == rhs.maxFramerate &&
lhs.qpMax == rhs.qpMax &&
lhs.numberOfSimulcastStreams == rhs.numberOfSimulcastStreams &&
lhs.mode == rhs.mode;
if (ret && lhs.codecType == webrtc::kVideoCodecVP8) {
ret &= (lhs.codecSpecific.VP8 == rhs.codecSpecific.VP8);
}
for (unsigned char i = 0; i < rhs.numberOfSimulcastStreams && ret; ++i) {
ret &= (lhs.simulcastStream[i] == rhs.simulcastStream[i]);
}
return ret;
}
bool operator!=(const webrtc::VideoCodec& lhs,
const webrtc::VideoCodec& rhs) {
return !(lhs == rhs);
}
#endif
struct FlushBlackFrameData : public talk_base::MessageData {
FlushBlackFrameData(uint32 s, int64 t) : ssrc(s), timestamp(t) {
}
uint32 ssrc;
int64 timestamp;
};
class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
public:
explicit WebRtcRenderAdapter(VideoRenderer* renderer)
: renderer_(renderer), width_(0), height_(0) {
}
virtual ~WebRtcRenderAdapter() {
}
void SetRenderer(VideoRenderer* renderer) {
talk_base::CritScope cs(&crit_);
renderer_ = renderer;
// FrameSizeChange may have already been called when renderer was not set.
// If so we should call SetSize here.
// TODO(ronghuawu): Add unit test for this case. Didn't do it now
// because the WebRtcRenderAdapter is currently hiding in cc file. No
// good way to get access to it from the unit test.
if (width_ > 0 && height_ > 0 && renderer_ != NULL) {
if (!renderer_->SetSize(width_, height_, 0)) {
LOG(LS_ERROR)
<< "WebRtcRenderAdapter SetRenderer failed to SetSize to: "
<< width_ << "x" << height_;
}
}
}
// Implementation of webrtc::ExternalRenderer.
virtual int FrameSizeChange(unsigned int width, unsigned int height,
unsigned int /*number_of_streams*/) {
talk_base::CritScope cs(&crit_);
width_ = width;
height_ = height;
LOG(LS_INFO) << "WebRtcRenderAdapter frame size changed to: "
<< width << "x" << height;
if (renderer_ == NULL) {
LOG(LS_VERBOSE) << "WebRtcRenderAdapter the renderer has not been set. "
<< "SetSize will be called later in SetRenderer.";
return 0;
}
return renderer_->SetSize(width_, height_, 0) ? 0 : -1;
}
virtual int DeliverFrame(unsigned char* buffer, int buffer_size,
uint32_t time_stamp, int64_t render_time,
void* handle) {
talk_base::CritScope cs(&crit_);
frame_rate_tracker_.Update(1);
if (renderer_ == NULL) {
return 0;
}
// Convert 90K rtp timestamp to ns timestamp.
int64 rtp_time_stamp_in_ns = (time_stamp / 90) *
talk_base::kNumNanosecsPerMillisec;
// Convert milisecond render time to ns timestamp.
int64 render_time_stamp_in_ns = render_time *
talk_base::kNumNanosecsPerMillisec;
// Send the rtp timestamp to renderer as the VideoFrame timestamp.
// and the render timestamp as the VideoFrame elapsed_time.
if (handle == NULL) {
return DeliverBufferFrame(buffer, buffer_size, render_time_stamp_in_ns,
rtp_time_stamp_in_ns);
} else {
return DeliverTextureFrame(handle, render_time_stamp_in_ns,
rtp_time_stamp_in_ns);
}
}
virtual bool IsTextureSupported() { return true; }
int DeliverBufferFrame(unsigned char* buffer, int buffer_size,
int64 elapsed_time, int64 time_stamp) {
WebRtcVideoFrame video_frame;
video_frame.Alias(buffer, buffer_size, width_, height_,
1, 1, elapsed_time, time_stamp, 0);
// Sanity check on decoded frame size.
if (buffer_size != static_cast<int>(VideoFrame::SizeOf(width_, height_))) {
LOG(LS_WARNING) << "WebRtcRenderAdapter received a strange frame size: "
<< buffer_size;
}
int ret = renderer_->RenderFrame(&video_frame) ? 0 : -1;
return ret;
}
int DeliverTextureFrame(void* handle, int64 elapsed_time, int64 time_stamp) {
WebRtcTextureVideoFrame video_frame(
static_cast<webrtc::NativeHandle*>(handle), width_, height_,
elapsed_time, time_stamp);
return renderer_->RenderFrame(&video_frame);
}
unsigned int width() {
talk_base::CritScope cs(&crit_);
return width_;
}
unsigned int height() {
talk_base::CritScope cs(&crit_);
return height_;
}
int framerate() {
talk_base::CritScope cs(&crit_);
return static_cast<int>(frame_rate_tracker_.units_second());
}
VideoRenderer* renderer() {
talk_base::CritScope cs(&crit_);
return renderer_;
}
private:
talk_base::CriticalSection crit_;
VideoRenderer* renderer_;
unsigned int width_;
unsigned int height_;
talk_base::RateTracker frame_rate_tracker_;
};
class WebRtcDecoderObserver : public webrtc::ViEDecoderObserver {
public:
explicit WebRtcDecoderObserver(int video_channel)
: video_channel_(video_channel),
framerate_(0),
bitrate_(0),
decode_ms_(0),
max_decode_ms_(0),
current_delay_ms_(0),
target_delay_ms_(0),
jitter_buffer_ms_(0),
min_playout_delay_ms_(0),
render_delay_ms_(0) {
}
// virtual functions from VieDecoderObserver.
virtual void IncomingCodecChanged(const int videoChannel,
const webrtc::VideoCodec& videoCodec) {}
virtual void IncomingRate(const int videoChannel,
const unsigned int framerate,
const unsigned int bitrate) {
talk_base::CritScope cs(&crit_);
ASSERT(video_channel_ == videoChannel);
framerate_ = framerate;
bitrate_ = bitrate;
}
virtual void DecoderTiming(int decode_ms,
int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
int jitter_buffer_ms,
int min_playout_delay_ms,
int render_delay_ms) {
talk_base::CritScope cs(&crit_);
decode_ms_ = decode_ms;
max_decode_ms_ = max_decode_ms;
current_delay_ms_ = current_delay_ms;
target_delay_ms_ = target_delay_ms;
jitter_buffer_ms_ = jitter_buffer_ms;
min_playout_delay_ms_ = min_playout_delay_ms;
render_delay_ms_ = render_delay_ms;
}
virtual void RequestNewKeyFrame(const int videoChannel) {}
// Populate |rinfo| based on previously-set data in |*this|.
void ExportTo(VideoReceiverInfo* rinfo) {
talk_base::CritScope cs(&crit_);
rinfo->framerate_rcvd = framerate_;
rinfo->decode_ms = decode_ms_;
rinfo->max_decode_ms = max_decode_ms_;
rinfo->current_delay_ms = current_delay_ms_;
rinfo->target_delay_ms = target_delay_ms_;
rinfo->jitter_buffer_ms = jitter_buffer_ms_;
rinfo->min_playout_delay_ms = min_playout_delay_ms_;
rinfo->render_delay_ms = render_delay_ms_;
}
private:
mutable talk_base::CriticalSection crit_;
int video_channel_;
int framerate_;
int bitrate_;
int decode_ms_;
int max_decode_ms_;
int current_delay_ms_;
int target_delay_ms_;
int jitter_buffer_ms_;
int min_playout_delay_ms_;
int render_delay_ms_;
};
class WebRtcEncoderObserver : public webrtc::ViEEncoderObserver {
public:
explicit WebRtcEncoderObserver(int video_channel)
: video_channel_(video_channel),
framerate_(0),
bitrate_(0),
suspended_(false) {
}
// virtual functions from VieEncoderObserver.
virtual void OutgoingRate(const int videoChannel,
const unsigned int framerate,
const unsigned int bitrate) {
talk_base::CritScope cs(&crit_);
ASSERT(video_channel_ == videoChannel);
framerate_ = framerate;
bitrate_ = bitrate;
}
virtual void SuspendChange(int video_channel, bool is_suspended) {
talk_base::CritScope cs(&crit_);
ASSERT(video_channel_ == video_channel);
suspended_ = is_suspended;
}
int framerate() const {
talk_base::CritScope cs(&crit_);
return framerate_;
}
int bitrate() const {
talk_base::CritScope cs(&crit_);
return bitrate_;
}
bool suspended() const {
talk_base::CritScope cs(&crit_);
return suspended_;
}
private:
mutable talk_base::CriticalSection crit_;
int video_channel_;
int framerate_;
int bitrate_;
bool suspended_;
};
class WebRtcLocalStreamInfo {
public:
WebRtcLocalStreamInfo()
: width_(0), height_(0), elapsed_time_(-1), time_stamp_(-1) {}
size_t width() const {
talk_base::CritScope cs(&crit_);
return width_;
}
size_t height() const {
talk_base::CritScope cs(&crit_);
return height_;
}
int64 elapsed_time() const {
talk_base::CritScope cs(&crit_);
return elapsed_time_;
}
int64 time_stamp() const {
talk_base::CritScope cs(&crit_);
return time_stamp_;
}
int framerate() {
talk_base::CritScope cs(&crit_);
return static_cast<int>(rate_tracker_.units_second());
}
void GetLastFrameInfo(
size_t* width, size_t* height, int64* elapsed_time) const {
talk_base::CritScope cs(&crit_);
*width = width_;
*height = height_;
*elapsed_time = elapsed_time_;
}
void UpdateFrame(const VideoFrame* frame) {
talk_base::CritScope cs(&crit_);
width_ = frame->GetWidth();
height_ = frame->GetHeight();
elapsed_time_ = frame->GetElapsedTime();
time_stamp_ = frame->GetTimeStamp();
rate_tracker_.Update(1);
}
private:
mutable talk_base::CriticalSection crit_;
size_t width_;
size_t height_;
int64 elapsed_time_;
int64 time_stamp_;
talk_base::RateTracker rate_tracker_;
DISALLOW_COPY_AND_ASSIGN(WebRtcLocalStreamInfo);
};
// WebRtcVideoChannelRecvInfo is a container class with members such as renderer
// and a decoder observer that is used by receive channels.
// It must exist as long as the receive channel is connected to renderer or a
// decoder observer in this class and methods in the class should only be called
// from the worker thread.
class WebRtcVideoChannelRecvInfo {
public:
typedef std::map<int, webrtc::VideoDecoder*> DecoderMap; // key: payload type
explicit WebRtcVideoChannelRecvInfo(int channel_id)
: channel_id_(channel_id),
render_adapter_(NULL),
decoder_observer_(channel_id) {
}
int channel_id() { return channel_id_; }
void SetRenderer(VideoRenderer* renderer) {
render_adapter_.SetRenderer(renderer);
}
WebRtcRenderAdapter* render_adapter() { return &render_adapter_; }
WebRtcDecoderObserver* decoder_observer() { return &decoder_observer_; }
void RegisterDecoder(int pl_type, webrtc::VideoDecoder* decoder) {
ASSERT(!IsDecoderRegistered(pl_type));
registered_decoders_[pl_type] = decoder;
}
bool IsDecoderRegistered(int pl_type) {
return registered_decoders_.count(pl_type) != 0;
}
const DecoderMap& registered_decoders() {
return registered_decoders_;
}
void ClearRegisteredDecoders() {
registered_decoders_.clear();
}
private:
int channel_id_; // Webrtc video channel number.
// Renderer for this channel.
WebRtcRenderAdapter render_adapter_;
WebRtcDecoderObserver decoder_observer_;
DecoderMap registered_decoders_;
};
class WebRtcOveruseObserver : public webrtc::CpuOveruseObserver {
public:
explicit WebRtcOveruseObserver(CoordinatedVideoAdapter* video_adapter)
: video_adapter_(video_adapter),
enabled_(false) {
}
// TODO(mflodman): Consider sending resolution as part of event, to let
// adapter know what resolution the request is based on. Helps eliminate stale
// data, race conditions.
virtual void OveruseDetected() OVERRIDE {
talk_base::CritScope cs(&crit_);
if (!enabled_) {
return;
}
video_adapter_->OnCpuResolutionRequest(CoordinatedVideoAdapter::DOWNGRADE);
}
virtual void NormalUsage() OVERRIDE {
talk_base::CritScope cs(&crit_);
if (!enabled_) {
return;
}
video_adapter_->OnCpuResolutionRequest(CoordinatedVideoAdapter::UPGRADE);
}
void Enable(bool enable) {
talk_base::CritScope cs(&crit_);
enabled_ = enable;
}
bool enabled() const { return enabled_; }
private:
CoordinatedVideoAdapter* video_adapter_;
bool enabled_;
talk_base::CriticalSection crit_;
};
class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
public:
typedef std::map<int, webrtc::VideoEncoder*> EncoderMap; // key: payload type
WebRtcVideoChannelSendInfo(int channel_id, int capture_id,
webrtc::ViEExternalCapture* external_capture,
talk_base::CpuMonitor* cpu_monitor)
: channel_id_(channel_id),
capture_id_(capture_id),
sending_(false),
muted_(false),
video_capturer_(NULL),
encoder_observer_(channel_id),
external_capture_(external_capture),
capturer_updated_(false),
interval_(0),
cpu_monitor_(cpu_monitor),
overuse_observer_enabled_(false) {
}
int channel_id() const { return channel_id_; }
int capture_id() const { return capture_id_; }
void set_sending(bool sending) { sending_ = sending; }
bool sending() const { return sending_; }
void set_muted(bool on) {
// TODO(asapersson): add support.
// video_adapter_.SetBlackOutput(on);
muted_ = on;
}
bool muted() {return muted_; }
WebRtcEncoderObserver* encoder_observer() { return &encoder_observer_; }
webrtc::ViEExternalCapture* external_capture() { return external_capture_; }
const VideoFormat& video_format() const {
return video_format_;
}
void set_video_format(const VideoFormat& video_format) {
video_format_ = video_format;
if (video_format_ != cricket::VideoFormat()) {
interval_ = video_format_.interval;
}
CoordinatedVideoAdapter* adapter = video_adapter();
if (adapter) {
adapter->OnOutputFormatRequest(video_format_);
}
}
void set_interval(int64 interval) {
if (video_format() == cricket::VideoFormat()) {
interval_ = interval;
}
}
int64 interval() { return interval_; }
int CurrentAdaptReason() const {
const CoordinatedVideoAdapter* adapter = video_adapter();
if (!adapter) {
return CoordinatedVideoAdapter::ADAPTREASON_NONE;
}
return video_adapter()->adapt_reason();
}
StreamParams* stream_params() { return stream_params_.get(); }
void set_stream_params(const StreamParams& sp) {
stream_params_.reset(new StreamParams(sp));
}
void ClearStreamParams() { stream_params_.reset(); }
bool has_ssrc(uint32 local_ssrc) const {
return !stream_params_ ? false :
stream_params_->has_ssrc(local_ssrc);
}
WebRtcLocalStreamInfo* local_stream_info() {
return &local_stream_info_;
}
VideoCapturer* video_capturer() {
return video_capturer_;
}
void set_video_capturer(VideoCapturer* video_capturer,
ViEWrapper* vie_wrapper) {
if (video_capturer == video_capturer_) {
return;
}
CoordinatedVideoAdapter* old_video_adapter = video_adapter();
if (old_video_adapter) {
// Disconnect signals from old video adapter.
SignalCpuAdaptationUnable.disconnect(old_video_adapter);
if (cpu_monitor_) {
cpu_monitor_->SignalUpdate.disconnect(old_video_adapter);
}
}
capturer_updated_ = true;
video_capturer_ = video_capturer;
vie_wrapper->base()->RegisterCpuOveruseObserver(channel_id_, NULL);
if (!video_capturer) {
overuse_observer_.reset();
return;
}
CoordinatedVideoAdapter* adapter = video_adapter();
ASSERT(adapter && "Video adapter should not be null here.");
UpdateAdapterCpuOptions();
overuse_observer_.reset(new WebRtcOveruseObserver(adapter));
vie_wrapper->base()->RegisterCpuOveruseObserver(channel_id_,
overuse_observer_.get());
// (Dis)connect the video adapter from the cpu monitor as appropriate.
SetCpuOveruseDetection(overuse_observer_enabled_);
SignalCpuAdaptationUnable.repeat(adapter->SignalCpuAdaptationUnable);
}
CoordinatedVideoAdapter* video_adapter() {
if (!video_capturer_) {
return NULL;
}
return video_capturer_->video_adapter();
}
const CoordinatedVideoAdapter* video_adapter() const {
if (!video_capturer_) {
return NULL;
}
return video_capturer_->video_adapter();
}
void ApplyCpuOptions(const VideoOptions& video_options) {
// Use video_options_.SetAll() instead of assignment so that unset value in
// video_options will not overwrite the previous option value.
video_options_.SetAll(video_options);
UpdateAdapterCpuOptions();
}
void UpdateAdapterCpuOptions() {
if (!video_capturer_) {
return;
}
bool cpu_adapt, cpu_smoothing, adapt_third;
float low, med, high;
// TODO(thorcarpenter): Have VideoAdapter be responsible for setting
// all these video options.
CoordinatedVideoAdapter* video_adapter = video_capturer_->video_adapter();
if (video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt) ||
overuse_observer_enabled_) {
video_adapter->set_cpu_adaptation(cpu_adapt || overuse_observer_enabled_);
}
if (video_options_.adapt_cpu_with_smoothing.Get(&cpu_smoothing)) {
video_adapter->set_cpu_smoothing(cpu_smoothing);
}
if (video_options_.process_adaptation_threshhold.Get(&med)) {
video_adapter->set_process_threshold(med);
}
if (video_options_.system_low_adaptation_threshhold.Get(&low)) {
video_adapter->set_low_system_threshold(low);
}
if (video_options_.system_high_adaptation_threshhold.Get(&high)) {
video_adapter->set_high_system_threshold(high);
}
if (video_options_.video_adapt_third.Get(&adapt_third)) {
video_adapter->set_scale_third(adapt_third);
}
}
void SetCpuOveruseDetection(bool enable) {
overuse_observer_enabled_ = enable;
if (overuse_observer_) {
overuse_observer_->Enable(enable);
}
// The video adapter is signaled by overuse detection if enabled; otherwise
// it will be signaled by cpu monitor.
CoordinatedVideoAdapter* adapter = video_adapter();
if (adapter) {
bool cpu_adapt = false;
video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt);
adapter->set_cpu_adaptation(
adapter->cpu_adaptation() || cpu_adapt || enable);
if (cpu_monitor_) {
if (enable) {
cpu_monitor_->SignalUpdate.disconnect(adapter);
} else {
cpu_monitor_->SignalUpdate.connect(
adapter, &CoordinatedVideoAdapter::OnCpuLoadUpdated);
}
}
}
}
void ProcessFrame(const VideoFrame& original_frame, bool mute,
VideoFrame** processed_frame) {
if (!mute) {
*processed_frame = original_frame.Copy();
} else {
WebRtcVideoFrame* black_frame = new WebRtcVideoFrame();
black_frame->InitToBlack(static_cast<int>(original_frame.GetWidth()),
static_cast<int>(original_frame.GetHeight()),
1, 1,
original_frame.GetElapsedTime(),
original_frame.GetTimeStamp());
*processed_frame = black_frame;
}
local_stream_info_.UpdateFrame(*processed_frame);
}
void RegisterEncoder(int pl_type, webrtc::VideoEncoder* encoder) {
ASSERT(!IsEncoderRegistered(pl_type));
registered_encoders_[pl_type] = encoder;
}
bool IsEncoderRegistered(int pl_type) {
return registered_encoders_.count(pl_type) != 0;
}
const EncoderMap& registered_encoders() {
return registered_encoders_;
}
void ClearRegisteredEncoders() {
registered_encoders_.clear();
}
sigslot::repeater0<> SignalCpuAdaptationUnable;
private:
int channel_id_;
int capture_id_;
bool sending_;
bool muted_;
VideoCapturer* video_capturer_;
WebRtcEncoderObserver encoder_observer_;
webrtc::ViEExternalCapture* external_capture_;
EncoderMap registered_encoders_;
VideoFormat video_format_;
talk_base::scoped_ptr<StreamParams> stream_params_;
WebRtcLocalStreamInfo local_stream_info_;
bool capturer_updated_;
int64 interval_;
talk_base::CpuMonitor* cpu_monitor_;
talk_base::scoped_ptr<WebRtcOveruseObserver> overuse_observer_;
bool overuse_observer_enabled_;
VideoOptions video_options_;
};
const WebRtcVideoEngine::VideoCodecPref
WebRtcVideoEngine::kVideoCodecPrefs[] = {
{kVp8PayloadName, 100, -1, 0},
{kRedPayloadName, 116, -1, 1},
{kFecPayloadName, 117, -1, 2},
{kRtxCodecName, 96, 100, 3},
};
// The formats are sorted by the descending order of width. We use the order to
// find the next format for CPU and bandwidth adaptation.
const VideoFormatPod WebRtcVideoEngine::kVideoFormats[] = {
{1280, 800, FPS_TO_INTERVAL(30), FOURCC_ANY},
{1280, 720, FPS_TO_INTERVAL(30), FOURCC_ANY},
{960, 600, FPS_TO_INTERVAL(30), FOURCC_ANY},
{960, 540, FPS_TO_INTERVAL(30), FOURCC_ANY},
{640, 400, FPS_TO_INTERVAL(30), FOURCC_ANY},
{640, 360, FPS_TO_INTERVAL(30), FOURCC_ANY},
{640, 480, FPS_TO_INTERVAL(30), FOURCC_ANY},
{480, 300, FPS_TO_INTERVAL(30), FOURCC_ANY},
{480, 270, FPS_TO_INTERVAL(30), FOURCC_ANY},
{480, 360, FPS_TO_INTERVAL(30), FOURCC_ANY},
{320, 200, FPS_TO_INTERVAL(30), FOURCC_ANY},
{320, 180, FPS_TO_INTERVAL(30), FOURCC_ANY},
{320, 240, FPS_TO_INTERVAL(30), FOURCC_ANY},
{240, 150, FPS_TO_INTERVAL(30), FOURCC_ANY},
{240, 135, FPS_TO_INTERVAL(30), FOURCC_ANY},
{240, 180, FPS_TO_INTERVAL(30), FOURCC_ANY},
{160, 100, FPS_TO_INTERVAL(30), FOURCC_ANY},
{160, 90, FPS_TO_INTERVAL(30), FOURCC_ANY},
{160, 120, FPS_TO_INTERVAL(30), FOURCC_ANY},
};
const VideoFormatPod WebRtcVideoEngine::kDefaultVideoFormat =
{640, 400, FPS_TO_INTERVAL(30), FOURCC_ANY};
static void UpdateVideoCodec(const cricket::VideoFormat& video_format,
webrtc::VideoCodec* target_codec) {
if ((target_codec == NULL) || (video_format == cricket::VideoFormat())) {
return;
}
target_codec->width = video_format.width;
target_codec->height = video_format.height;
target_codec->maxFramerate = cricket::VideoFormat::IntervalToFps(
video_format.interval);
}
#ifdef USE_WEBRTC_DEV_BRANCH
static bool GetCpuOveruseOptions(const VideoOptions& options,
webrtc::CpuOveruseOptions* overuse_options) {
int underuse_threshold = 0;
int overuse_threshold = 0;
if (!options.cpu_underuse_threshold.Get(&underuse_threshold) ||
!options.cpu_overuse_threshold.Get(&overuse_threshold)) {
return false;
}
if (underuse_threshold <= 0 || overuse_threshold <= 0) {
return false;
}
// Valid thresholds.
bool encode_usage =
options.cpu_overuse_encode_usage.GetWithDefaultIfUnset(false);
overuse_options->enable_capture_jitter_method = !encode_usage;
overuse_options->enable_encode_usage_method = encode_usage;
if (encode_usage) {
// Use method based on encode usage.
overuse_options->low_encode_usage_threshold_percent = underuse_threshold;
overuse_options->high_encode_usage_threshold_percent = overuse_threshold;
} else {
// Use default method based on capture jitter.
overuse_options->low_capture_jitter_threshold_ms =
static_cast<float>(underuse_threshold);
overuse_options->high_capture_jitter_threshold_ms =
static_cast<float>(overuse_threshold);
}
return true;
}
#endif
WebRtcVideoEngine::WebRtcVideoEngine() {
Construct(new ViEWrapper(), new ViETraceWrapper(), NULL,
new talk_base::CpuMonitor(NULL));
}
WebRtcVideoEngine::WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
ViEWrapper* vie_wrapper,
talk_base::CpuMonitor* cpu_monitor) {
Construct(vie_wrapper, new ViETraceWrapper(), voice_engine, cpu_monitor);
}
WebRtcVideoEngine::WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing,
talk_base::CpuMonitor* cpu_monitor) {
Construct(vie_wrapper, tracing, voice_engine, cpu_monitor);
}
void WebRtcVideoEngine::Construct(ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing,
WebRtcVoiceEngine* voice_engine,
talk_base::CpuMonitor* cpu_monitor) {
LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine";
worker_thread_ = NULL;
vie_wrapper_.reset(vie_wrapper);
vie_wrapper_base_initialized_ = false;
tracing_.reset(tracing);
voice_engine_ = voice_engine;
initialized_ = false;
SetTraceFilter(SeverityToFilter(kDefaultLogSeverity));
render_module_.reset(new WebRtcPassthroughRender());
local_renderer_w_ = local_renderer_h_ = 0;
local_renderer_ = NULL;
capture_started_ = false;
decoder_factory_ = NULL;
encoder_factory_ = NULL;
cpu_monitor_.reset(cpu_monitor);
SetTraceOptions("");
if (tracing_->SetTraceCallback(this) != 0) {
LOG_RTCERR1(SetTraceCallback, this);
}
// Set default quality levels for our supported codecs. We override them here
// if we know your cpu performance is low, and they can be updated explicitly
// by calling SetDefaultCodec. For example by a flute preference setting, or
// by the server with a jec in response to our reported system info.
VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
kVideoCodecPrefs[0].name,
kDefaultVideoFormat.width,
kDefaultVideoFormat.height,
VideoFormat::IntervalToFps(kDefaultVideoFormat.interval),
0);
if (!SetDefaultCodec(max_codec)) {
LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
}
// Load our RTP Header extensions.
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
kRtpTimestampOffsetHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
}
WebRtcVideoEngine::~WebRtcVideoEngine() {
LOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine";
if (initialized_) {
Terminate();
}
if (encoder_factory_) {
encoder_factory_->RemoveObserver(this);
}
tracing_->SetTraceCallback(NULL);
// Test to see if the media processor was deregistered properly.
ASSERT(SignalMediaFrame.is_empty());
}
bool WebRtcVideoEngine::Init(talk_base::Thread* worker_thread) {
LOG(LS_INFO) << "WebRtcVideoEngine::Init";
worker_thread_ = worker_thread;
ASSERT(worker_thread_ != NULL);
cpu_monitor_->set_thread(worker_thread_);
if (!cpu_monitor_->Start(kCpuMonitorPeriodMs)) {
LOG(LS_ERROR) << "Failed to start CPU monitor.";
cpu_monitor_.reset();
}
bool result = InitVideoEngine();
if (result) {
LOG(LS_INFO) << "VideoEngine Init done";
} else {
LOG(LS_ERROR) << "VideoEngine Init failed, releasing";
Terminate();
}
return result;
}
bool WebRtcVideoEngine::InitVideoEngine() {
LOG(LS_INFO) << "WebRtcVideoEngine::InitVideoEngine";
// Init WebRTC VideoEngine.
if (!vie_wrapper_base_initialized_) {
if (vie_wrapper_->base()->Init() != 0) {
LOG_RTCERR0(Init);
return false;
}
vie_wrapper_base_initialized_ = true;
}
// Log the VoiceEngine version info.
char buffer[1024] = "";
if (vie_wrapper_->base()->GetVersion(buffer) != 0) {
LOG_RTCERR0(GetVersion);
return false;
}
LOG(LS_INFO) << "WebRtc VideoEngine Version:";
LogMultiline(talk_base::LS_INFO, buffer);
// Hook up to VoiceEngine for sync purposes, if supplied.
if (!voice_engine_) {
LOG(LS_WARNING) << "NULL voice engine";
} else if ((vie_wrapper_->base()->SetVoiceEngine(
voice_engine_->voe()->engine())) != 0) {
LOG_RTCERR0(SetVoiceEngine);
return false;
}
// Register our custom render module.
if (vie_wrapper_->render()->RegisterVideoRenderModule(
*render_module_.get()) != 0) {
LOG_RTCERR0(RegisterVideoRenderModule);
return false;
}
initialized_ = true;
return true;
}
void WebRtcVideoEngine::Terminate() {
LOG(LS_INFO) << "WebRtcVideoEngine::Terminate";
initialized_ = false;
if (vie_wrapper_->render()->DeRegisterVideoRenderModule(
*render_module_.get()) != 0) {
LOG_RTCERR0(DeRegisterVideoRenderModule);
}
if (vie_wrapper_->base()->SetVoiceEngine(NULL) != 0) {
LOG_RTCERR0(SetVoiceEngine);
}
cpu_monitor_->Stop();
}
int WebRtcVideoEngine::GetCapabilities() {
return VIDEO_RECV | VIDEO_SEND;
}
bool WebRtcVideoEngine::SetOptions(const VideoOptions &options) {
return true;
}
bool WebRtcVideoEngine::SetDefaultEncoderConfig(
const VideoEncoderConfig& config) {
return SetDefaultCodec(config.max_codec);
}
VideoEncoderConfig WebRtcVideoEngine::GetDefaultEncoderConfig() const {
ASSERT(!video_codecs_.empty());
VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
kVideoCodecPrefs[0].name,
video_codecs_[0].width,
video_codecs_[0].height,
video_codecs_[0].framerate,
0);
return VideoEncoderConfig(max_codec);
}
// SetDefaultCodec may be called while the capturer is running. For example, a
// test call is started in a page with QVGA default codec, and then a real call
// is started in another page with VGA default codec. This is the corner case
// and happens only when a session is started. We ignore this case currently.
bool WebRtcVideoEngine::SetDefaultCodec(const VideoCodec& codec) {
if (!RebuildCodecList(codec)) {
LOG(LS_WARNING) << "Failed to RebuildCodecList";
return false;
}
ASSERT(!video_codecs_.empty());
default_codec_format_ = VideoFormat(
video_codecs_[0].width,
video_codecs_[0].height,
VideoFormat::FpsToInterval(video_codecs_[0].framerate),
FOURCC_ANY);
return true;
}
WebRtcVideoMediaChannel* WebRtcVideoEngine::CreateChannel(
VoiceMediaChannel* voice_channel) {
WebRtcVideoMediaChannel* channel =
new WebRtcVideoMediaChannel(this, voice_channel);
if (!channel->Init()) {
delete channel;
channel = NULL;
}
return channel;
}
bool WebRtcVideoEngine::SetLocalRenderer(VideoRenderer* renderer) {
local_renderer_w_ = local_renderer_h_ = 0;
local_renderer_ = renderer;
return true;
}
const std::vector<VideoCodec>& WebRtcVideoEngine::codecs() const {
return video_codecs_;
}
const std::vector<RtpHeaderExtension>&
WebRtcVideoEngine::rtp_header_extensions() const {
return rtp_header_extensions_;
}
void WebRtcVideoEngine::SetLogging(int min_sev, const char* filter) {
// if min_sev == -1, we keep the current log level.
if (min_sev >= 0) {
SetTraceFilter(SeverityToFilter(min_sev));
}
SetTraceOptions(filter);
}
int WebRtcVideoEngine::GetLastEngineError() {
return vie_wrapper_->error();
}
// Checks to see whether we comprehend and could receive a particular codec
bool WebRtcVideoEngine::FindCodec(const VideoCodec& in) {
for (int i = 0; i < ARRAY_SIZE(kVideoFormats); ++i) {
const VideoFormat fmt(kVideoFormats[i]);
if ((in.width == 0 && in.height == 0) ||
(fmt.width == in.width && fmt.height == in.height)) {
if (encoder_factory_) {
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
encoder_factory_->codecs();
for (size_t j = 0; j < codecs.size(); ++j) {
VideoCodec codec(GetExternalVideoPayloadType(static_cast<int>(j)),
codecs[j].name, 0, 0, 0, 0);
if (codec.Matches(in))
return true;
}
}
for (size_t j = 0; j < ARRAY_SIZE(kVideoCodecPrefs); ++j) {
VideoCodec codec(kVideoCodecPrefs[j].payload_type,
kVideoCodecPrefs[j].name, 0, 0, 0, 0);
if (codec.Matches(in)) {
return true;
}
}
}
}
return false;
}
// Given the requested codec, returns true if we can send that codec type and
// updates out with the best quality we could send for that codec. If current is
// not empty, we constrain out so that its aspect ratio matches current's.
bool WebRtcVideoEngine::CanSendCodec(const VideoCodec& requested,
const VideoCodec& current,
VideoCodec* out) {
if (!out) {
return false;
}
std::vector<VideoCodec>::const_iterator local_max;
for (local_max = video_codecs_.begin();
local_max < video_codecs_.end();
++local_max) {
// First match codecs by payload type
if (!requested.Matches(*local_max)) {
continue;
}
out->id = requested.id;
out->name = requested.name;
out->preference = requested.preference;
out->params = requested.params;
out->framerate = talk_base::_min(requested.framerate, local_max->framerate);
out->width = 0;
out->height = 0;
out->params = requested.params;
out->feedback_params = requested.feedback_params;
if (0 == requested.width && 0 == requested.height) {
// Special case with resolution 0. The channel should not send frames.
return true;
} else if (0 == requested.width || 0 == requested.height) {
// 0xn and nx0 are invalid resolutions.
return false;
}
// Pick the best quality that is within their and our bounds and has the
// correct aspect ratio.
for (int j = 0; j < ARRAY_SIZE(kVideoFormats); ++j) {
const VideoFormat format(kVideoFormats[j]);
// Skip any format that is larger than the local or remote maximums, or
// smaller than the current best match
if (format.width > requested.width || format.height > requested.height ||
format.width > local_max->width ||
(format.width < out->width && format.height < out->height)) {
continue;
}
bool better = false;
// Check any further constraints on this prospective format
if (!out->width || !out->height) {
// If we don't have any matches yet, this is the best so far.
better = true;
} else if (current.width && current.height) {
// current is set so format must match its ratio exactly.
better =
(format.width * current.height == format.height * current.width);
} else {
// Prefer closer aspect ratios i.e
// format.aspect - requested.aspect < out.aspect - requested.aspect
better = abs(format.width * requested.height * out->height -
requested.width * format.height * out->height) <
abs(out->width * format.height * requested.height -
requested.width * format.height * out->height);
}
if (better) {
out->width = format.width;
out->height = format.height;
}
}
if (out->width > 0) {
return true;
}
}
return false;
}
static void ConvertToCricketVideoCodec(
const webrtc::VideoCodec& in_codec, VideoCodec* out_codec) {
out_codec->id = in_codec.plType;
out_codec->name = in_codec.plName;
out_codec->width = in_codec.width;
out_codec->height = in_codec.height;
out_codec->framerate = in_codec.maxFramerate;
out_codec->SetParam(kCodecParamMinBitrate, in_codec.minBitrate);
out_codec->SetParam(kCodecParamMaxBitrate, in_codec.maxBitrate);
if (in_codec.qpMax) {
out_codec->SetParam(kCodecParamMaxQuantization, in_codec.qpMax);
}
}
bool WebRtcVideoEngine::ConvertFromCricketVideoCodec(
const VideoCodec& in_codec, webrtc::VideoCodec* out_codec) {
bool found = false;
int ncodecs = vie_wrapper_->codec()->NumberOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
if (vie_wrapper_->codec()->GetCodec(i, *out_codec) == 0 &&
_stricmp(in_codec.name.c_str(), out_codec->plName) == 0) {
found = true;
break;
}
}
// If not found, check if this is supported by external encoder factory.
if (!found && encoder_factory_) {
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
encoder_factory_->codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
if (_stricmp(in_codec.name.c_str(), codecs[i].name.c_str()) == 0) {
out_codec->codecType = codecs[i].type;
out_codec->plType = GetExternalVideoPayloadType(static_cast<int>(i));
talk_base::strcpyn(out_codec->plName, sizeof(out_codec->plName),
codecs[i].name.c_str(), codecs[i].name.length());
found = true;
break;
}
}
}
if (!found) {
LOG(LS_ERROR) << "invalid codec type";
return false;
}
if (in_codec.id != 0)
out_codec->plType = in_codec.id;
if (in_codec.width != 0)
out_codec->width = in_codec.width;
if (in_codec.height != 0)
out_codec->height = in_codec.height;
if (in_codec.framerate != 0)
out_codec->maxFramerate = in_codec.framerate;
// Convert bitrate parameters.
int max_bitrate = kMaxVideoBitrate;
int min_bitrate = kMinVideoBitrate;
int start_bitrate = kStartVideoBitrate;
in_codec.GetParam(kCodecParamMinBitrate, &min_bitrate);
in_codec.GetParam(kCodecParamMaxBitrate, &max_bitrate);
if (max_bitrate < min_bitrate) {
return false;
}
start_bitrate = talk_base::_max(start_bitrate, min_bitrate);
start_bitrate = talk_base::_min(start_bitrate, max_bitrate);
out_codec->minBitrate = min_bitrate;
out_codec->startBitrate = start_bitrate;
out_codec->maxBitrate = max_bitrate;
// Convert general codec parameters.
int max_quantization = 0;
if (in_codec.GetParam(kCodecParamMaxQuantization, &max_quantization)) {
if (max_quantization < 0) {
return false;
}
out_codec->qpMax = max_quantization;
}
return true;
}
void WebRtcVideoEngine::RegisterChannel(WebRtcVideoMediaChannel *channel) {
talk_base::CritScope cs(&channels_crit_);
channels_.push_back(channel);
}
void WebRtcVideoEngine::UnregisterChannel(WebRtcVideoMediaChannel *channel) {
talk_base::CritScope cs(&channels_crit_);
channels_.erase(std::remove(channels_.begin(), channels_.end(), channel),
channels_.end());
}
bool WebRtcVideoEngine::SetVoiceEngine(WebRtcVoiceEngine* voice_engine) {
if (initialized_) {
LOG(LS_WARNING) << "SetVoiceEngine can not be called after Init";
return false;
}
voice_engine_ = voice_engine;
return true;
}
bool WebRtcVideoEngine::EnableTimedRender() {
if (initialized_) {
LOG(LS_WARNING) << "EnableTimedRender can not be called after Init";
return false;
}
render_module_.reset(webrtc::VideoRender::CreateVideoRender(0, NULL,
false, webrtc::kRenderExternal));
return true;
}
void WebRtcVideoEngine::SetTraceFilter(int filter) {
tracing_->SetTraceFilter(filter);
}
// See https://sites.google.com/a/google.com/wavelet/
// Home/Magic-Flute--RTC-Engine-/Magic-Flute-Command-Line-Parameters
// for all supported command line setttings.
void WebRtcVideoEngine::SetTraceOptions(const std::string& options) {
// Set WebRTC trace file.
std::vector<std::string> opts;
talk_base::tokenize(options, ' ', '"', '"', &opts);
std::vector<std::string>::iterator tracefile =
std::find(opts.begin(), opts.end(), "tracefile");
if (tracefile != opts.end() && ++tracefile != opts.end()) {
// Write WebRTC debug output (at same loglevel) to file
if (tracing_->SetTraceFile(tracefile->c_str()) == -1) {
LOG_RTCERR1(SetTraceFile, *tracefile);
}
}
}
static void AddDefaultFeedbackParams(VideoCodec* codec) {
const FeedbackParam kFir(kRtcpFbParamCcm, kRtcpFbCcmParamFir);
codec->AddFeedbackParam(kFir);
const FeedbackParam kNack(kRtcpFbParamNack, kParamValueEmpty);
codec->AddFeedbackParam(kNack);
const FeedbackParam kPli(kRtcpFbParamNack, kRtcpFbNackParamPli);
codec->AddFeedbackParam(kPli);
const FeedbackParam kRemb(kRtcpFbParamRemb, kParamValueEmpty);
codec->AddFeedbackParam(kRemb);
}
// Rebuilds the codec list to be only those that are less intensive
// than the specified codec. Prefers internal codec over external with
// higher preference field.
bool WebRtcVideoEngine::RebuildCodecList(const VideoCodec& in_codec) {
if (!FindCodec(in_codec))
return false;
video_codecs_.clear();
bool found = false;
std::set<std::string> internal_codec_names;
for (size_t i = 0; i < ARRAY_SIZE(kVideoCodecPrefs); ++i) {
const VideoCodecPref& pref(kVideoCodecPrefs[i]);
if (!found)
found = (in_codec.name == pref.name);
if (found) {
VideoCodec codec(pref.payload_type, pref.name,
in_codec.width, in_codec.height, in_codec.framerate,
static_cast<int>(ARRAY_SIZE(kVideoCodecPrefs) - i));
if (_stricmp(kVp8PayloadName, codec.name.c_str()) == 0) {
AddDefaultFeedbackParams(&codec);
}
if (pref.associated_payload_type != -1) {
codec.SetParam(kCodecParamAssociatedPayloadType,
pref.associated_payload_type);
}
video_codecs_.push_back(codec);
internal_codec_names.insert(codec.name);
}
}
if (encoder_factory_) {
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
encoder_factory_->codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
bool is_internal_codec = internal_codec_names.find(codecs[i].name) !=
internal_codec_names.end();
if (!is_internal_codec) {
if (!found)
found = (in_codec.name == codecs[i].name);
VideoCodec codec(
GetExternalVideoPayloadType(static_cast<int>(i)),
codecs[i].name,
codecs[i].max_width,
codecs[i].max_height,
codecs[i].max_fps,
// Use negative preference on external codec to ensure the internal
// codec is preferred.
static_cast<int>(0 - i));
AddDefaultFeedbackParams(&codec);
video_codecs_.push_back(codec);
}
}
}
ASSERT(found);
return true;
}
// Ignore spammy trace messages, mostly from the stats API when we haven't
// gotten RTCP info yet from the remote side.
bool WebRtcVideoEngine::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;
}
int WebRtcVideoEngine::GetNumOfChannels() {
talk_base::CritScope cs(&channels_crit_);
return static_cast<int>(channels_.size());
}
void WebRtcVideoEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
sev = talk_base::LS_ERROR;
else if (level == webrtc::kTraceWarning)
sev = talk_base::LS_WARNING;
else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
sev = talk_base::LS_INFO;
else if (level == webrtc::kTraceTerseInfo)
sev = talk_base::LS_INFO;
// Skip past boilerplate prefix text
if (length < 72) {
std::string msg(trace, length);
LOG(LS_ERROR) << "Malformed webrtc log message: ";
LOG_V(sev) << msg;
} else {
std::string msg(trace + 71, length - 72);
if (!ShouldIgnoreTrace(msg) &&
(!voice_engine_ || !voice_engine_->ShouldIgnoreTrace(msg))) {
LOG_V(sev) << "webrtc: " << msg;
}
}
}
webrtc::VideoDecoder* WebRtcVideoEngine::CreateExternalDecoder(
webrtc::VideoCodecType type) {
if (decoder_factory_ == NULL) {
return NULL;
}
return decoder_factory_->CreateVideoDecoder(type);
}
void WebRtcVideoEngine::DestroyExternalDecoder(webrtc::VideoDecoder* decoder) {
ASSERT(decoder_factory_ != NULL);
if (decoder_factory_ == NULL)
return;
decoder_factory_->DestroyVideoDecoder(decoder);
}
webrtc::VideoEncoder* WebRtcVideoEngine::CreateExternalEncoder(
webrtc::VideoCodecType type) {
if (encoder_factory_ == NULL) {
return NULL;
}
return encoder_factory_->CreateVideoEncoder(type);
}
void WebRtcVideoEngine::DestroyExternalEncoder(webrtc::VideoEncoder* encoder) {
ASSERT(encoder_factory_ != NULL);
if (encoder_factory_ == NULL)
return;
encoder_factory_->DestroyVideoEncoder(encoder);
}
bool WebRtcVideoEngine::IsExternalEncoderCodecType(
webrtc::VideoCodecType type) const {
if (!encoder_factory_)
return false;
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
encoder_factory_->codecs();
std::vector<WebRtcVideoEncoderFactory::VideoCodec>::const_iterator it;
for (it = codecs.begin(); it != codecs.end(); ++it) {
if (it->type == type)
return true;
}
return false;
}
void WebRtcVideoEngine::SetExternalDecoderFactory(
WebRtcVideoDecoderFactory* decoder_factory) {
decoder_factory_ = decoder_factory;
}
void WebRtcVideoEngine::SetExternalEncoderFactory(
WebRtcVideoEncoderFactory* encoder_factory) {
if (encoder_factory_ == encoder_factory)
return;
if (encoder_factory_) {
encoder_factory_->RemoveObserver(this);
}
encoder_factory_ = encoder_factory;
if (encoder_factory_) {
encoder_factory_->AddObserver(this);
}
// Invoke OnCodecAvailable() here in case the list of codecs is already
// available when the encoder factory is installed. If not the encoder
// factory will invoke the callback later when the codecs become available.
OnCodecsAvailable();
}
void WebRtcVideoEngine::OnCodecsAvailable() {
// Rebuild codec list while reapplying the current default codec format.
VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
kVideoCodecPrefs[0].name,
video_codecs_[0].width,
video_codecs_[0].height,
video_codecs_[0].framerate,
0);
if (!RebuildCodecList(max_codec)) {
LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
}
}
// WebRtcVideoMediaChannel
WebRtcVideoMediaChannel::WebRtcVideoMediaChannel(
WebRtcVideoEngine* engine,
VoiceMediaChannel* channel)
: engine_(engine),
voice_channel_(channel),
vie_channel_(-1),
nack_enabled_(true),
remb_enabled_(false),
render_started_(false),
first_receive_ssrc_(0),
num_unsignalled_recv_channels_(0),
send_rtx_type_(-1),
send_red_type_(-1),
send_fec_type_(-1),
send_min_bitrate_(kMinVideoBitrate),
send_start_bitrate_(kStartVideoBitrate),
send_max_bitrate_(kMaxVideoBitrate),
sending_(false),
ratio_w_(0),
ratio_h_(0) {
engine->RegisterChannel(this);
}
bool WebRtcVideoMediaChannel::Init() {
const uint32 ssrc_key = 0;
return CreateChannel(ssrc_key, MD_SENDRECV, &vie_channel_);
}
WebRtcVideoMediaChannel::~WebRtcVideoMediaChannel() {
const bool send = false;
SetSend(send);
const bool render = false;
SetRender(render);
while (!send_channels_.empty()) {
if (!DeleteSendChannel(send_channels_.begin()->first)) {
LOG(LS_ERROR) << "Unable to delete channel with ssrc key "
<< send_channels_.begin()->first;
ASSERT(false);
break;
}
}
// Remove all receive streams and the default channel.
while (!recv_channels_.empty()) {
RemoveRecvStreamInternal(recv_channels_.begin()->first);
}
// Unregister the channel from the engine.
engine()->UnregisterChannel(this);
if (worker_thread()) {
worker_thread()->Clear(this);
}
}
bool WebRtcVideoMediaChannel::SetRecvCodecs(
const std::vector<VideoCodec>& codecs) {
receive_codecs_.clear();
for (std::vector<VideoCodec>::const_iterator iter = codecs.begin();
iter != codecs.end(); ++iter) {
if (engine()->FindCodec(*iter)) {
webrtc::VideoCodec wcodec;
if (engine()->ConvertFromCricketVideoCodec(*iter, &wcodec)) {
receive_codecs_.push_back(wcodec);
}
} else {
LOG(LS_INFO) << "Unknown codec " << iter->name;
return false;
}
}
for (RecvChannelMap::iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
if (!SetReceiveCodecs(it->second))
return false;
}
return true;
}
bool WebRtcVideoMediaChannel::SetSendCodecs(
const std::vector<VideoCodec>& codecs) {
// Match with local video codec list.
std::vector<webrtc::VideoCodec> send_codecs;
VideoCodec checked_codec;
VideoCodec current; // defaults to 0x0
if (sending_) {
ConvertToCricketVideoCodec(*send_codec_, &current);
}
std::map<int, int> primary_rtx_pt_mapping;
bool nack_enabled = nack_enabled_;
bool remb_enabled = remb_enabled_;
for (std::vector<VideoCodec>::const_iterator iter = codecs.begin();
iter != codecs.end(); ++iter) {
if (_stricmp(iter->name.c_str(), kRedPayloadName) == 0) {
send_red_type_ = iter->id;
} else if (_stricmp(iter->name.c_str(), kFecPayloadName) == 0) {
send_fec_type_ = iter->id;
} else if (_stricmp(iter->name.c_str(), kRtxCodecName) == 0) {
int rtx_type = iter->id;
int rtx_primary_type = -1;
if (iter->GetParam(kCodecParamAssociatedPayloadType, &rtx_primary_type)) {
primary_rtx_pt_mapping[rtx_primary_type] = rtx_type;
}
} else if (engine()->CanSendCodec(*iter, current, &checked_codec)) {
webrtc::VideoCodec wcodec;
if (engine()->ConvertFromCricketVideoCodec(checked_codec, &wcodec)) {
if (send_codecs.empty()) {
nack_enabled = IsNackEnabled(checked_codec);
remb_enabled = IsRembEnabled(checked_codec);
}
send_codecs.push_back(wcodec);
}
} else {
LOG(LS_WARNING) << "Unknown codec " << iter->name;
}
}
// Fail if we don't have a match.
if (send_codecs.empty()) {
LOG(LS_WARNING) << "No matching codecs available";
return false;
}
// Recv protection.
// Do not update if the status is same as previously configured.
if (nack_enabled_ != nack_enabled) {
for (RecvChannelMap::iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
int channel_id = it->second->channel_id();
if (!SetNackFec(channel_id, send_red_type_, send_fec_type_,
nack_enabled)) {
return false;
}
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
kNotSending,
remb_enabled_) != 0) {
LOG_RTCERR3(SetRembStatus, channel_id, kNotSending, remb_enabled_);
return false;
}
}
nack_enabled_ = nack_enabled;
}
// Send settings.
// Do not update if the status is same as previously configured.
if (remb_enabled_ != remb_enabled) {
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
int channel_id = iter->second->channel_id();
if (!SetNackFec(channel_id, send_red_type_, send_fec_type_,
nack_enabled_)) {
return false;
}
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
remb_enabled,
remb_enabled) != 0) {
LOG_RTCERR3(SetRembStatus, channel_id, remb_enabled, remb_enabled);
return false;
}
}
remb_enabled_ = remb_enabled;
}
// Select the first matched codec.
webrtc::VideoCodec& codec(send_codecs[0]);
// Set RTX payload type if primary now active. This value will be used in
// SetSendCodec.
std::map<int, int>::const_iterator rtx_it =
primary_rtx_pt_mapping.find(static_cast<int>(codec.plType));
if (rtx_it != primary_rtx_pt_mapping.end()) {
send_rtx_type_ = rtx_it->second;
}
if (!SetSendCodec(
codec, codec.minBitrate, codec.startBitrate, codec.maxBitrate)) {
return false;
}
LogSendCodecChange("SetSendCodecs()");
return true;
}
bool WebRtcVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) {
if (!send_codec_) {
return false;
}
ConvertToCricketVideoCodec(*send_codec_, send_codec);
return true;
}
bool WebRtcVideoMediaChannel::SetSendStreamFormat(uint32 ssrc,
const VideoFormat& format) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(ssrc);
if (!send_channel) {
LOG(LS_ERROR) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
send_channel->set_video_format(format);
return true;
}
bool WebRtcVideoMediaChannel::SetRender(bool render) {
if (render == render_started_) {
return true; // no action required
}
bool ret = true;
for (RecvChannelMap::iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
if (render) {
if (engine()->vie()->render()->StartRender(
it->second->channel_id()) != 0) {
LOG_RTCERR1(StartRender, it->second->channel_id());
ret = false;
}
} else {
if (engine()->vie()->render()->StopRender(
it->second->channel_id()) != 0) {
LOG_RTCERR1(StopRender, it->second->channel_id());
ret = false;
}
}
}
if (ret) {
render_started_ = render;
}
return ret;
}
bool WebRtcVideoMediaChannel::SetSend(bool send) {
if (!HasReadySendChannels() && send) {
LOG(LS_ERROR) << "No stream added";
return false;
}
if (send == sending()) {
return true; // No action required.
}
if (send) {
// We've been asked to start sending.
// SetSendCodecs must have been called already.
if (!send_codec_) {
return false;
}
// Start send now.
if (!StartSend()) {
return false;
}
} else {
// We've been asked to stop sending.
if (!StopSend()) {
return false;
}
}
sending_ = send;
return true;
}
bool WebRtcVideoMediaChannel::AddSendStream(const StreamParams& sp) {
if (sp.first_ssrc() == 0) {
LOG(LS_ERROR) << "AddSendStream with 0 ssrc is not supported.";
return false;
}
LOG(LS_INFO) << "AddSendStream " << sp.ToString();
if (!IsOneSsrcStream(sp) && !IsSimulcastStream(sp)) {
LOG(LS_ERROR) << "AddSendStream: bad local stream parameters";
return false;
}
uint32 ssrc_key;
if (!CreateSendChannelKey(sp.first_ssrc(), &ssrc_key)) {
LOG(LS_ERROR) << "Trying to register duplicate ssrc: " << sp.first_ssrc();
return false;
}
// If the default channel is already used for sending create a new channel
// otherwise use the default channel for sending.
int channel_id = -1;
if (send_channels_[0]->stream_params() == NULL) {
channel_id = vie_channel_;
} else {
if (!CreateChannel(ssrc_key, MD_SEND, &channel_id)) {
LOG(LS_ERROR) << "AddSendStream: unable to create channel";
return false;
}
}
WebRtcVideoChannelSendInfo* send_channel = send_channels_[ssrc_key];
// Set the send (local) SSRC.
// If there are multiple send SSRCs, we can only set the first one here, and
// the rest of the SSRC(s) need to be set after SetSendCodec has been called
// (with a codec requires multiple SSRC(s)).
if (engine()->vie()->rtp()->SetLocalSSRC(channel_id,
sp.first_ssrc()) != 0) {
LOG_RTCERR2(SetLocalSSRC, channel_id, sp.first_ssrc());
return false;
}
// Set the corresponding RTX SSRC.
if (!SetLocalRtxSsrc(channel_id, sp, sp.first_ssrc(), 0)) {
return false;
}
// Set RTCP CName.
if (engine()->vie()->rtp()->SetRTCPCName(channel_id,
sp.cname.c_str()) != 0) {
LOG_RTCERR2(SetRTCPCName, channel_id, sp.cname.c_str());
return false;
}
// At this point the channel's local SSRC has been updated. If the channel is
// the default channel make sure that all the receive channels are updated as
// well. Receive channels have to have the same SSRC as the default channel in
// order to send receiver reports with this SSRC.
if (IsDefaultChannel(channel_id)) {
for (RecvChannelMap::const_iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
WebRtcVideoChannelRecvInfo* info = it->second;
int channel_id = info->channel_id();
if (engine()->vie()->rtp()->SetLocalSSRC(channel_id,
sp.first_ssrc()) != 0) {
LOG_RTCERR1(SetLocalSSRC, it->first);
return false;
}
}
}
send_channel->set_stream_params(sp);
// Reset send codec after stream parameters changed.
if (send_codec_) {
if (!SetSendCodec(send_channel, *send_codec_, send_min_bitrate_,
send_start_bitrate_, send_max_bitrate_)) {
return false;
}
LogSendCodecChange("SetSendStreamFormat()");
}
if (sending_) {
return StartSend(send_channel);
}
return true;
}
bool WebRtcVideoMediaChannel::RemoveSendStream(uint32 ssrc) {
if (ssrc == 0) {
LOG(LS_ERROR) << "RemoveSendStream with 0 ssrc is not supported.";
return false;
}
uint32 ssrc_key;
if (!GetSendChannelKey(ssrc, &ssrc_key)) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
WebRtcVideoChannelSendInfo* send_channel = send_channels_[ssrc_key];
int channel_id = send_channel->channel_id();
if (IsDefaultChannel(channel_id) && (send_channel->stream_params() == NULL)) {
// Default channel will still exist. However, if stream_params() is NULL
// there is no stream to remove.
return false;
}
if (sending_) {
StopSend(send_channel);
}
const WebRtcVideoChannelSendInfo::EncoderMap& encoder_map =
send_channel->registered_encoders();
for (WebRtcVideoChannelSendInfo::EncoderMap::const_iterator it =
encoder_map.begin(); it != encoder_map.end(); ++it) {
if (engine()->vie()->ext_codec()->DeRegisterExternalSendCodec(
channel_id, it->first) != 0) {
LOG_RTCERR1(DeregisterEncoderObserver, channel_id);
}
engine()->DestroyExternalEncoder(it->second);
}
send_channel->ClearRegisteredEncoders();
// The receive channels depend on the default channel, recycle it instead.
if (IsDefaultChannel(channel_id)) {
SetCapturer(GetDefaultChannelSsrc(), NULL);
send_channel->ClearStreamParams();
} else {
return DeleteSendChannel(ssrc_key);
}
return true;
}
bool WebRtcVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
if (sp.first_ssrc() == 0) {
LOG(LS_ERROR) << "AddRecvStream with 0 ssrc is not supported.";
return false;
}
// TODO(zhurunz) Remove this once BWE works properly across different send
// and receive channels.
// Reuse default channel for recv stream in 1:1 call.
if (!InConferenceMode() && first_receive_ssrc_ == 0) {
LOG(LS_INFO) << "Recv stream " << sp.first_ssrc()
<< " reuse default channel #"
<< vie_channel_;
first_receive_ssrc_ = sp.first_ssrc();
if (render_started_) {
if (engine()->vie()->render()->StartRender(vie_channel_) !=0) {
LOG_RTCERR1(StartRender, vie_channel_);
}
}
return true;
}
int channel_id = -1;
RecvChannelMap::iterator channel_iterator =
recv_channels_.find(sp.first_ssrc());
if (channel_iterator == recv_channels_.end() &&
first_receive_ssrc_ != sp.first_ssrc()) {
// TODO(perkj): Implement recv media from multiple media SSRCs per stream.
// NOTE: We have two SSRCs per stream when RTX is enabled.
if (!IsOneSsrcStream(sp)) {
LOG(LS_ERROR) << "WebRtcVideoMediaChannel supports one primary SSRC per"
<< " stream and one FID SSRC per primary SSRC.";
return false;
}
// Create a new channel for receiving video data.
// In order to get the bandwidth estimation work fine for
// receive only channels, we connect all receiving channels
// to our master send channel.
if (!CreateChannel(sp.first_ssrc(), MD_RECV, &channel_id)) {
return false;
}
} else {
// Already exists.
if (first_receive_ssrc_ == sp.first_ssrc()) {
return false;
}
// Early receive added channel.
channel_id = (*channel_iterator).second->channel_id();
}
// Set the corresponding RTX SSRC.
uint32 rtx_ssrc;
bool has_rtx = sp.GetFidSsrc(sp.first_ssrc(), &rtx_ssrc);
if (has_rtx && engine()->vie()->rtp()->SetRemoteSSRCType(
channel_id, webrtc::kViEStreamTypeRtx, rtx_ssrc) != 0) {
LOG_RTCERR3(SetRemoteSSRCType, channel_id, webrtc::kViEStreamTypeRtx,
rtx_ssrc);
return false;
}
// Get the default renderer.
VideoRenderer* default_renderer = NULL;
if (InConferenceMode()) {
// The recv_channels_ size start out being 1, so if it is two here this
// is the first receive channel created (vie_channel_ is not used for
// receiving in a conference call). This means that the renderer stored
// inside vie_channel_ should be used for the just created channel.
if (recv_channels_.size() == 2 &&
recv_channels_.find(0) != recv_channels_.end()) {
GetRenderer(0, &default_renderer);
}
}
// The first recv stream reuses the default renderer (if a default renderer
// has been set).
if (default_renderer) {
SetRenderer(sp.first_ssrc(), default_renderer);
}
LOG(LS_INFO) << "New video stream " << sp.first_ssrc()
<< " registered to VideoEngine channel #"
<< channel_id << " and connected to channel #" << vie_channel_;
return true;
}
bool WebRtcVideoMediaChannel::RemoveRecvStream(uint32 ssrc) {
if (ssrc == 0) {
LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
return false;
}
return RemoveRecvStreamInternal(ssrc);
}
bool WebRtcVideoMediaChannel::RemoveRecvStreamInternal(uint32 ssrc) {
RecvChannelMap::iterator it = recv_channels_.find(ssrc);
if (it == recv_channels_.end()) {
// TODO(perkj): Remove this once BWE works properly across different send
// and receive channels.
// The default channel is reused for recv stream in 1:1 call.
if (first_receive_ssrc_ == ssrc) {
first_receive_ssrc_ = 0;
// Need to stop the renderer and remove it since the render window can be
// deleted after this.
if (render_started_) {
if (engine()->vie()->render()->StopRender(vie_channel_) !=0) {
LOG_RTCERR1(StopRender, it->second->channel_id());
}
}
recv_channels_[0]->SetRenderer(NULL);
return true;
}
return false;
}
WebRtcVideoChannelRecvInfo* info = it->second;
int channel_id = info->channel_id();
if (engine()->vie()->render()->RemoveRenderer(channel_id) != 0) {
LOG_RTCERR1(RemoveRenderer, channel_id);
}
if (engine()->vie()->network()->DeregisterSendTransport(channel_id) !=0) {
LOG_RTCERR1(DeRegisterSendTransport, channel_id);
}
if (engine()->vie()->codec()->DeregisterDecoderObserver(
channel_id) != 0) {
LOG_RTCERR1(DeregisterDecoderObserver, channel_id);
}
const WebRtcVideoChannelRecvInfo::DecoderMap& decoder_map =
info->registered_decoders();
for (WebRtcVideoChannelRecvInfo::DecoderMap::const_iterator it =
decoder_map.begin(); it != decoder_map.end(); ++it) {
if (engine()->vie()->ext_codec()->DeRegisterExternalReceiveCodec(
channel_id, it->first) != 0) {
LOG_RTCERR1(DeregisterDecoderObserver, channel_id);
}
engine()->DestroyExternalDecoder(it->second);
}
info->ClearRegisteredDecoders();
LOG(LS_INFO) << "Removing video stream " << ssrc
<< " with VideoEngine channel #"
<< channel_id;
bool ret = true;
if (engine()->vie()->base()->DeleteChannel(channel_id) == -1) {
LOG_RTCERR1(DeleteChannel, channel_id);
ret = false;
}
// Delete the WebRtcVideoChannelRecvInfo pointed to by it->second.
delete info;
recv_channels_.erase(it);
return ret;
}
bool WebRtcVideoMediaChannel::StartSend() {
bool success = true;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (!StartSend(send_channel)) {
success = false;
}
}
return success;
}
bool WebRtcVideoMediaChannel::StartSend(
WebRtcVideoChannelSendInfo* send_channel) {
const int channel_id = send_channel->channel_id();
if (engine()->vie()->base()->StartSend(channel_id) != 0) {
LOG_RTCERR1(StartSend, channel_id);
return false;
}
send_channel->set_sending(true);
return true;
}
bool WebRtcVideoMediaChannel::StopSend() {
bool success = true;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (!StopSend(send_channel)) {
success = false;
}
}
return success;
}
bool WebRtcVideoMediaChannel::StopSend(
WebRtcVideoChannelSendInfo* send_channel) {
const int channel_id = send_channel->channel_id();
if (engine()->vie()->base()->StopSend(channel_id) != 0) {
LOG_RTCERR1(StopSend, channel_id);
return false;
}
send_channel->set_sending(false);
return true;
}
bool WebRtcVideoMediaChannel::SendIntraFrame() {
bool success = true;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end();
++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
const int channel_id = send_channel->channel_id();
if (engine()->vie()->codec()->SendKeyFrame(channel_id) != 0) {
LOG_RTCERR1(SendKeyFrame, channel_id);
success = false;
}
}
return success;
}
bool WebRtcVideoMediaChannel::HasReadySendChannels() {
return !send_channels_.empty() &&
((send_channels_.size() > 1) ||
(send_channels_[0]->stream_params() != NULL));
}
bool WebRtcVideoMediaChannel::GetSendChannelKey(uint32 local_ssrc,
uint32* key) {
*key = 0;
// If a send channel is not ready to send it will not have local_ssrc
// registered to it.
if (!HasReadySendChannels()) {
return false;
}
// The default channel is stored with key 0. The key therefore does not match
// the SSRC associated with the default channel. Check if the SSRC provided
// corresponds to the default channel's SSRC.
if (local_ssrc == GetDefaultChannelSsrc()) {
return true;
}
if (send_channels_.find(local_ssrc) == send_channels_.end()) {
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->has_ssrc(local_ssrc)) {
*key = iter->first;
return true;
}
}
return false;
}
// The key was found in the above std::map::find call. This means that the
// ssrc is the key.
*key = local_ssrc;
return true;
}
WebRtcVideoChannelSendInfo* WebRtcVideoMediaChannel::GetSendChannel(
uint32 local_ssrc) {
uint32 key;
if (!GetSendChannelKey(local_ssrc, &key)) {
return NULL;
}
return send_channels_[key];
}
bool WebRtcVideoMediaChannel::CreateSendChannelKey(uint32 local_ssrc,
uint32* key) {
if (GetSendChannelKey(local_ssrc, key)) {
// If there is a key corresponding to |local_ssrc|, the SSRC is already in
// use. SSRCs need to be unique in a session and at this point a duplicate
// SSRC has been detected.
return false;
}
if (send_channels_[0]->stream_params() == NULL) {
// key should be 0 here as the default channel should be re-used whenever it
// is not used.
*key = 0;
return true;
}
// SSRC is currently not in use and the default channel is already in use. Use
// the SSRC as key since it is supposed to be unique in a session.
*key = local_ssrc;
return true;
}
int WebRtcVideoMediaChannel::GetSendChannelNum(VideoCapturer* capturer) {
int num = 0;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == capturer) {
++num;
}
}
return num;
}
uint32 WebRtcVideoMediaChannel::GetDefaultChannelSsrc() {
WebRtcVideoChannelSendInfo* send_channel = send_channels_[0];
const StreamParams* sp = send_channel->stream_params();
if (sp == NULL) {
// This happens if no send stream is currently registered.
return 0;
}
return sp->first_ssrc();
}
bool WebRtcVideoMediaChannel::DeleteSendChannel(uint32 ssrc_key) {
if (send_channels_.find(ssrc_key) == send_channels_.end()) {
return false;
}
WebRtcVideoChannelSendInfo* send_channel = send_channels_[ssrc_key];
MaybeDisconnectCapturer(send_channel->video_capturer());
send_channel->set_video_capturer(NULL, engine()->vie());
int channel_id = send_channel->channel_id();
int capture_id = send_channel->capture_id();
if (engine()->vie()->codec()->DeregisterEncoderObserver(
channel_id) != 0) {
LOG_RTCERR1(DeregisterEncoderObserver, channel_id);
}
// Destroy the external capture interface.
if (engine()->vie()->capture()->DisconnectCaptureDevice(
channel_id) != 0) {
LOG_RTCERR1(DisconnectCaptureDevice, channel_id);
}
if (engine()->vie()->capture()->ReleaseCaptureDevice(
capture_id) != 0) {
LOG_RTCERR1(ReleaseCaptureDevice, capture_id);
}
// The default channel is stored in both |send_channels_| and
// |recv_channels_|. To make sure it is only deleted once from vie let the
// delete call happen when tearing down |recv_channels_| and not here.
if (!IsDefaultChannel(channel_id)) {
engine_->vie()->base()->DeleteChannel(channel_id);
}
delete send_channel;
send_channels_.erase(ssrc_key);
return true;
}
bool WebRtcVideoMediaChannel::RemoveCapturer(uint32 ssrc) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(ssrc);
if (!send_channel) {
return false;
}
VideoCapturer* capturer = send_channel->video_capturer();
if (capturer == NULL) {
return false;
}
MaybeDisconnectCapturer(capturer);
send_channel->set_video_capturer(NULL, engine()->vie());
const int64 timestamp = send_channel->local_stream_info()->time_stamp();
if (send_codec_) {
QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate);
}
return true;
}
bool WebRtcVideoMediaChannel::SetRenderer(uint32 ssrc,
VideoRenderer* renderer) {
if (recv_channels_.find(ssrc) == recv_channels_.end()) {
// TODO(perkj): Remove this once BWE works properly across different send
// and receive channels.
// The default channel is reused for recv stream in 1:1 call.
if (first_receive_ssrc_ == ssrc &&
recv_channels_.find(0) != recv_channels_.end()) {
LOG(LS_INFO) << "SetRenderer " << ssrc
<< " reuse default channel #"
<< vie_channel_;
recv_channels_[0]->SetRenderer(renderer);
return true;
}
return false;
}
recv_channels_[ssrc]->SetRenderer(renderer);
return true;
}
bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options,
VideoMediaInfo* info) {
// Get sender statistics and build VideoSenderInfo.
unsigned int total_bitrate_sent = 0;
unsigned int video_bitrate_sent = 0;
unsigned int fec_bitrate_sent = 0;
unsigned int nack_bitrate_sent = 0;
unsigned int estimated_send_bandwidth = 0;
unsigned int target_enc_bitrate = 0;
if (send_codec_) {
for (SendChannelMap::const_iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
const int channel_id = send_channel->channel_id();
VideoSenderInfo sinfo;
const StreamParams* send_params = send_channel->stream_params();
if (send_params == NULL) {
// This should only happen if the default vie channel is not in use.
// This can happen if no streams have ever been added or the stream
// corresponding to the default channel has been removed. Note that
// there may be non-default vie channels in use when this happen so
// asserting send_channels_.size() == 1 is not correct and neither is
// breaking out of the loop.
ASSERT(channel_id == vie_channel_);
continue;
}
unsigned int bytes_sent, packets_sent, bytes_recv, packets_recv;
if (engine_->vie()->rtp()->GetRTPStatistics(channel_id, bytes_sent,
packets_sent, bytes_recv,
packets_recv) != 0) {
LOG_RTCERR1(GetRTPStatistics, vie_channel_);
continue;
}
WebRtcLocalStreamInfo* channel_stream_info =
send_channel->local_stream_info();
for (size_t i = 0; i < send_params->ssrcs.size(); ++i) {
sinfo.add_ssrc(send_params->ssrcs[i]);
}
sinfo.codec_name = send_codec_->plName;
sinfo.bytes_sent = bytes_sent;
sinfo.packets_sent = packets_sent;
sinfo.packets_cached = -1;
sinfo.packets_lost = -1;
sinfo.fraction_lost = -1;
sinfo.rtt_ms = -1;
sinfo.input_frame_width = static_cast<int>(channel_stream_info->width());
sinfo.input_frame_height =
static_cast<int>(channel_stream_info->height());
VideoCapturer* video_capturer = send_channel->video_capturer();
if (video_capturer) {
video_capturer->GetStats(&sinfo.adapt_frame_drops,
&sinfo.effects_frame_drops,
&sinfo.capturer_frame_time);
}
webrtc::VideoCodec vie_codec;
// TODO(ronghuawu): Add unit tests to cover the new send stats:
// send_frame_width/height.
if (!video_capturer || video_capturer->IsMuted()) {
sinfo.send_frame_width = 0;
sinfo.send_frame_height = 0;
} else if (engine()->vie()->codec()->GetSendCodec(channel_id,
vie_codec) == 0) {
sinfo.send_frame_width = vie_codec.width;
sinfo.send_frame_height = vie_codec.height;
} else {
sinfo.send_frame_width = -1;
sinfo.send_frame_height = -1;
LOG_RTCERR1(GetSendCodec, channel_id);
}
sinfo.framerate_input = channel_stream_info->framerate();
sinfo.framerate_sent = send_channel->encoder_observer()->framerate();
sinfo.nominal_bitrate = send_channel->encoder_observer()->bitrate();
sinfo.preferred_bitrate = send_max_bitrate_;
sinfo.adapt_reason = send_channel->CurrentAdaptReason();
sinfo.capture_jitter_ms = -1;
sinfo.avg_encode_ms = -1;
sinfo.encode_usage_percent = -1;
sinfo.capture_queue_delay_ms_per_s = -1;
int capture_jitter_ms = 0;
int avg_encode_time_ms = 0;
int encode_usage_percent = 0;
int capture_queue_delay_ms_per_s = 0;
if (engine()->vie()->base()->CpuOveruseMeasures(
channel_id,
&capture_jitter_ms,
&avg_encode_time_ms,
&encode_usage_percent,
&capture_queue_delay_ms_per_s) == 0) {
sinfo.capture_jitter_ms = capture_jitter_ms;
sinfo.avg_encode_ms = avg_encode_time_ms;
sinfo.encode_usage_percent = encode_usage_percent;
sinfo.capture_queue_delay_ms_per_s = capture_queue_delay_ms_per_s;
}
#ifdef USE_WEBRTC_DEV_BRANCH
webrtc::RtcpPacketTypeCounter rtcp_sent;
webrtc::RtcpPacketTypeCounter rtcp_received;
if (engine()->vie()->rtp()->GetRtcpPacketTypeCounters(
channel_id, &rtcp_sent, &rtcp_received) == 0) {
sinfo.firs_rcvd = rtcp_received.fir_packets;
sinfo.plis_rcvd = rtcp_received.pli_packets;
sinfo.nacks_rcvd = rtcp_received.nack_packets;
} else {
sinfo.firs_rcvd = -1;
sinfo.plis_rcvd = -1;
sinfo.nacks_rcvd = -1;
LOG_RTCERR1(GetRtcpPacketTypeCounters, channel_id);
}
#else
sinfo.firs_rcvd = -1;
sinfo.plis_rcvd = -1;
sinfo.nacks_rcvd = -1;
#endif
// Get received RTCP statistics for the sender (reported by the remote
// client in a RTCP packet), if available.
// It's not a fatal error if we can't, since RTCP may not have arrived
// yet.
webrtc::RtcpStatistics outgoing_stream_rtcp_stats;
int outgoing_stream_rtt_ms;
if (engine_->vie()->rtp()->GetSendChannelRtcpStatistics(
channel_id,
outgoing_stream_rtcp_stats,
outgoing_stream_rtt_ms) == 0) {
// Convert Q8 to float.
sinfo.packets_lost = outgoing_stream_rtcp_stats.cumulative_lost;
sinfo.fraction_lost = static_cast<float>(
outgoing_stream_rtcp_stats.fraction_lost) / (1 << 8);
sinfo.rtt_ms = outgoing_stream_rtt_ms;
}
info->senders.push_back(sinfo);
unsigned int channel_total_bitrate_sent = 0;
unsigned int channel_video_bitrate_sent = 0;
unsigned int channel_fec_bitrate_sent = 0;
unsigned int channel_nack_bitrate_sent = 0;
if (engine_->vie()->rtp()->GetBandwidthUsage(
channel_id, channel_total_bitrate_sent, channel_video_bitrate_sent,
channel_fec_bitrate_sent, channel_nack_bitrate_sent) == 0) {
total_bitrate_sent += channel_total_bitrate_sent;
video_bitrate_sent += channel_video_bitrate_sent;
fec_bitrate_sent += channel_fec_bitrate_sent;
nack_bitrate_sent += channel_nack_bitrate_sent;
} else {
LOG_RTCERR1(GetBandwidthUsage, channel_id);
}
unsigned int estimated_stream_send_bandwidth = 0;
if (engine_->vie()->rtp()->GetEstimatedSendBandwidth(
channel_id, &estimated_stream_send_bandwidth) == 0) {
estimated_send_bandwidth += estimated_stream_send_bandwidth;
} else {
LOG_RTCERR1(GetEstimatedSendBandwidth, channel_id);
}
unsigned int target_enc_stream_bitrate = 0;
if (engine_->vie()->codec()->GetCodecTargetBitrate(
channel_id, &target_enc_stream_bitrate) == 0) {
target_enc_bitrate += target_enc_stream_bitrate;
} else {
LOG_RTCERR1(GetCodecTargetBitrate, channel_id);
}
}
} else {
LOG(LS_WARNING) << "GetStats: sender information not ready.";
}
// Get the SSRC and stats for each receiver, based on our own calculations.
unsigned int estimated_recv_bandwidth = 0;
for (RecvChannelMap::const_iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
WebRtcVideoChannelRecvInfo* channel = it->second;
unsigned int ssrc;
// Get receiver statistics and build VideoReceiverInfo, if we have data.
// Skip the default channel (ssrc == 0).
if (engine_->vie()->rtp()->GetRemoteSSRC(
channel->channel_id(), ssrc) != 0 ||
ssrc == 0)
continue;
webrtc::StreamDataCounters sent;
webrtc::StreamDataCounters received;
if (engine_->vie()->rtp()->GetRtpStatistics(channel->channel_id(),
sent, received) != 0) {
LOG_RTCERR1(GetRTPStatistics, channel->channel_id());
return false;
}
VideoReceiverInfo rinfo;
rinfo.add_ssrc(ssrc);
rinfo.bytes_rcvd = received.bytes;
rinfo.packets_rcvd = received.packets;
rinfo.packets_lost = -1;
rinfo.packets_concealed = -1;
rinfo.fraction_lost = -1; // from SentRTCP
rinfo.frame_width = channel->render_adapter()->width();
rinfo.frame_height = channel->render_adapter()->height();
int fps = channel->render_adapter()->framerate();
rinfo.framerate_decoded = fps;
rinfo.framerate_output = fps;
channel->decoder_observer()->ExportTo(&rinfo);
#ifdef USE_WEBRTC_DEV_BRANCH
webrtc::RtcpPacketTypeCounter rtcp_sent;
webrtc::RtcpPacketTypeCounter rtcp_received;
if (engine()->vie()->rtp()->GetRtcpPacketTypeCounters(
channel->channel_id(), &rtcp_sent, &rtcp_received) == 0) {
rinfo.firs_sent = rtcp_sent.fir_packets;
rinfo.plis_sent = rtcp_sent.pli_packets;
rinfo.nacks_sent = rtcp_sent.nack_packets;
} else {
rinfo.firs_sent = -1;
rinfo.plis_sent = -1;
rinfo.nacks_sent = -1;
LOG_RTCERR1(GetRtcpPacketTypeCounters, channel->channel_id());
}
#else
rinfo.firs_sent = -1;
rinfo.plis_sent = -1;
rinfo.nacks_sent = -1;
#endif
// Get our locally created statistics of the received RTP stream.
webrtc::RtcpStatistics incoming_stream_rtcp_stats;
int incoming_stream_rtt_ms;
if (engine_->vie()->rtp()->GetReceiveChannelRtcpStatistics(
channel->channel_id(),
incoming_stream_rtcp_stats,
incoming_stream_rtt_ms) == 0) {
// Convert Q8 to float.
rinfo.packets_lost = incoming_stream_rtcp_stats.cumulative_lost;
rinfo.fraction_lost = static_cast<float>(
incoming_stream_rtcp_stats.fraction_lost) / (1 << 8);
}
info->receivers.push_back(rinfo);
unsigned int estimated_recv_stream_bandwidth = 0;
if (engine_->vie()->rtp()->GetEstimatedReceiveBandwidth(
channel->channel_id(), &estimated_recv_stream_bandwidth) == 0) {
estimated_recv_bandwidth += estimated_recv_stream_bandwidth;
} else {
LOG_RTCERR1(GetEstimatedReceiveBandwidth, channel->channel_id());
}
}
// Build BandwidthEstimationInfo.
// TODO(zhurunz): Add real unittest for this.
BandwidthEstimationInfo bwe;
// TODO(jiayl): remove the condition when the necessary changes are available
// outside the dev branch.
#ifdef USE_WEBRTC_DEV_BRANCH
if (options.include_received_propagation_stats) {
webrtc::ReceiveBandwidthEstimatorStats additional_stats;
// Only call for the default channel because the returned stats are
// collected for all the channels using the same estimator.
if (engine_->vie()->rtp()->GetReceiveBandwidthEstimatorStats(
recv_channels_[0]->channel_id(), &additional_stats) == 0) {
bwe.total_received_propagation_delta_ms =
additional_stats.total_propagation_time_delta_ms;
bwe.recent_received_propagation_delta_ms.swap(
additional_stats.recent_propagation_time_delta_ms);
bwe.recent_received_packet_group_arrival_time_ms.swap(
additional_stats.recent_arrival_time_ms);
}
}
engine_->vie()->rtp()->GetPacerQueuingDelayMs(
recv_channels_[0]->channel_id(), &bwe.bucket_delay);
#endif
// Calculations done above per send/receive stream.
bwe.actual_enc_bitrate = video_bitrate_sent;
bwe.transmit_bitrate = total_bitrate_sent;
bwe.retransmit_bitrate = nack_bitrate_sent;
bwe.available_send_bandwidth = estimated_send_bandwidth;
bwe.available_recv_bandwidth = estimated_recv_bandwidth;
bwe.target_enc_bitrate = target_enc_bitrate;
info->bw_estimations.push_back(bwe);
return true;
}
bool WebRtcVideoMediaChannel::SetCapturer(uint32 ssrc,
VideoCapturer* capturer) {
ASSERT(ssrc != 0);
if (!capturer) {
return RemoveCapturer(ssrc);
}
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(ssrc);
if (!send_channel) {
return false;
}
VideoCapturer* old_capturer = send_channel->video_capturer();
MaybeDisconnectCapturer(old_capturer);
send_channel->set_video_capturer(capturer, engine()->vie());
MaybeConnectCapturer(capturer);
if (!capturer->IsScreencast() && ratio_w_ != 0 && ratio_h_ != 0) {
capturer->UpdateAspectRatio(ratio_w_, ratio_h_);
}
const int64 timestamp = send_channel->local_stream_info()->time_stamp();
if (send_codec_) {
QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate);
}
return true;
}
bool WebRtcVideoMediaChannel::RequestIntraFrame() {
// There is no API exposed to application to request a key frame
// ViE does this internally when there are errors from decoder
return false;
}
void WebRtcVideoMediaChannel::OnPacketReceived(
talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
// Pick which channel to send this packet to. If this packet doesn't match
// any multiplexed streams, just send it to the default channel. Otherwise,
// send it to the specific decoder instance for that stream.
uint32 ssrc = 0;
if (!GetRtpSsrc(packet->data(), packet->length(), &ssrc))
return;
int processing_channel = GetRecvChannelNum(ssrc);
if (processing_channel == -1) {
// Allocate an unsignalled recv channel for processing in conference mode.
if (!InConferenceMode()) {
// If we cant find or allocate one, use the default.
processing_channel = video_channel();
} else if (!CreateUnsignalledRecvChannel(ssrc, &processing_channel)) {
// If we cant create an unsignalled recv channel, drop the packet in
// conference mode.
return;
}
}
engine()->vie()->network()->ReceivedRTPPacket(
processing_channel,
packet->data(),
static_cast<int>(packet->length()),
webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
void WebRtcVideoMediaChannel::OnRtcpReceived(
talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
// Sending channels need all RTCP packets with feedback information.
// Even sender reports can contain attached report blocks.
// Receiving channels need sender reports in order to create
// correct receiver reports.
uint32 ssrc = 0;
if (!GetRtcpSsrc(packet->data(), packet->length(), &ssrc)) {
LOG(LS_WARNING) << "Failed to parse SSRC from received RTCP packet";
return;
}
int type = 0;
if (!GetRtcpType(packet->data(), packet->length(), &type)) {
LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
return;
}
// If it is a sender report, find the channel that is listening.
if (type == kRtcpTypeSR) {
int which_channel = GetRecvChannelNum(ssrc);
if (which_channel != -1 && !IsDefaultChannel(which_channel)) {
engine_->vie()->network()->ReceivedRTCPPacket(
which_channel,
packet->data(),
static_cast<int>(packet->length()));
}
}
// SR may continue RR and any RR entry may correspond to any one of the send
// channels. So all RTCP packets must be forwarded all send channels. ViE
// will filter out RR internally.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
int channel_id = send_channel->channel_id();
engine_->vie()->network()->ReceivedRTCPPacket(
channel_id,
packet->data(),
static_cast<int>(packet->length()));
}
}
void WebRtcVideoMediaChannel::OnReadyToSend(bool ready) {
SetNetworkTransmissionState(ready);
}
bool WebRtcVideoMediaChannel::MuteStream(uint32 ssrc, bool muted) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(ssrc);
if (!send_channel) {
LOG(LS_ERROR) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
send_channel->set_muted(muted);
return true;
}
bool WebRtcVideoMediaChannel::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
if (receive_extensions_ == extensions) {
return true;
}
receive_extensions_ = extensions;
const RtpHeaderExtension* offset_extension =
FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
// Loop through all receive channels and enable/disable the extensions.
for (RecvChannelMap::iterator channel_it = recv_channels_.begin();
channel_it != recv_channels_.end(); ++channel_it) {
int channel_id = channel_it->second->channel_id();
if (!SetHeaderExtension(
&webrtc::ViERTP_RTCP::SetReceiveTimestampOffsetStatus, channel_id,
offset_extension)) {
return false;
}
if (!SetHeaderExtension(
&webrtc::ViERTP_RTCP::SetReceiveAbsoluteSendTimeStatus, channel_id,
send_time_extension)) {
return false;
}
}
return true;
}
bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
send_extensions_ = extensions;
const RtpHeaderExtension* offset_extension =
FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
// Loop through all send channels and enable/disable the extensions.
for (SendChannelMap::iterator channel_it = send_channels_.begin();
channel_it != send_channels_.end(); ++channel_it) {
int channel_id = channel_it->second->channel_id();
if (!SetHeaderExtension(
&webrtc::ViERTP_RTCP::SetSendTimestampOffsetStatus, channel_id,
offset_extension)) {
return false;
}
if (!SetHeaderExtension(
&webrtc::ViERTP_RTCP::SetSendAbsoluteSendTimeStatus, channel_id,
send_time_extension)) {
return false;
}
}
if (send_time_extension) {
// For video RTP packets, we would like to update AbsoluteSendTimeHeader
// Extension closer to the network, @ socket level before sending.
// Pushing the extension id to socket layer.
MediaChannel::SetOption(NetworkInterface::ST_RTP,
talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID,
send_time_extension->id);
}
return true;
}
int WebRtcVideoMediaChannel::GetRtpSendTimeExtnId() const {
const RtpHeaderExtension* send_time_extension = FindHeaderExtension(
send_extensions_, kRtpAbsoluteSenderTimeHeaderExtension);
if (send_time_extension) {
return send_time_extension->id;
}
return -1;
}
bool WebRtcVideoMediaChannel::SetStartSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVideoMediaChannel::SetStartSendBandwidth";
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet";
return true;
}
// On success, SetSendCodec() will reset |send_start_bitrate_| to |bps/1000|,
// by calling MaybeChangeStartBitrate. That method will also clamp the
// start bitrate between min and max, consistent with the override behavior
// in SetMaxSendBandwidth.
return SetSendCodec(*send_codec_,
send_min_bitrate_, bps / 1000, send_max_bitrate_);
}
bool WebRtcVideoMediaChannel::SetMaxSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVideoMediaChannel::SetMaxSendBandwidth";
if (InConferenceMode()) {
LOG(LS_INFO) << "Conference mode ignores SetMaxSendBandwidth";
return true;
}
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet";
return true;
}
// Use the default value or the bps for the max
int max_bitrate = (bps <= 0) ? send_max_bitrate_ : (bps / 1000);
// Reduce the current minimum and start bitrates if necessary.
int min_bitrate = talk_base::_min(send_min_bitrate_, max_bitrate);
int start_bitrate = talk_base::_min(send_start_bitrate_, max_bitrate);
if (!SetSendCodec(*send_codec_, min_bitrate, start_bitrate, max_bitrate)) {
return false;
}
LogSendCodecChange("SetMaxSendBandwidth()");
return true;
}
bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
// Always accept options that are unchanged.
if (options_ == options) {
return true;
}
// Trigger SetSendCodec to set correct noise reduction state if the option has
// changed.
bool denoiser_changed = options.video_noise_reduction.IsSet() &&
(options_.video_noise_reduction != options.video_noise_reduction);
bool leaky_bucket_changed = options.video_leaky_bucket.IsSet() &&
(options_.video_leaky_bucket != options.video_leaky_bucket);
bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
(options_.buffered_mode_latency != options.buffered_mode_latency);
bool cpu_overuse_detection_changed = options.cpu_overuse_detection.IsSet() &&
(options_.cpu_overuse_detection != options.cpu_overuse_detection);
bool dscp_option_changed = (options_.dscp != options.dscp);
bool suspend_below_min_bitrate_changed =
options.suspend_below_min_bitrate.IsSet() &&
(options_.suspend_below_min_bitrate != options.suspend_below_min_bitrate);
bool conference_mode_turned_off = false;
if (options_.conference_mode.IsSet() && options.conference_mode.IsSet() &&
options_.conference_mode.GetWithDefaultIfUnset(false) &&
!options.conference_mode.GetWithDefaultIfUnset(false)) {
conference_mode_turned_off = true;
}
#ifdef USE_WEBRTC_DEV_BRANCH
bool improved_wifi_bwe_changed =
options.use_improved_wifi_bandwidth_estimator.IsSet() &&
options_.use_improved_wifi_bandwidth_estimator !=
options.use_improved_wifi_bandwidth_estimator;
#endif
// Save the options, to be interpreted where appropriate.
// Use options_.SetAll() instead of assignment so that unset value in options
// will not overwrite the previous option value.
options_.SetAll(options);
// Set CPU options for all send channels.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
send_channel->ApplyCpuOptions(options_);
}
// Adjust send codec bitrate if needed.
int conf_max_bitrate = kDefaultConferenceModeMaxVideoBitrate;
// Save altered min_bitrate level and apply if necessary.
bool adjusted_min_bitrate = false;
if (options.lower_min_bitrate.IsSet()) {
bool lower;
options.lower_min_bitrate.Get(&lower);
int new_send_min_bitrate = lower ? kLowerMinBitrate : kMinVideoBitrate;
adjusted_min_bitrate = (new_send_min_bitrate != send_min_bitrate_);
send_min_bitrate_ = new_send_min_bitrate;
}
int expected_bitrate = send_max_bitrate_;
if (InConferenceMode()) {
expected_bitrate = conf_max_bitrate;
} else if (conference_mode_turned_off) {
// This is a special case for turning conference mode off.
// Max bitrate should go back to the default maximum value instead
// of the current maximum.
expected_bitrate = kMaxVideoBitrate;
}
int options_start_bitrate;
bool start_bitrate_changed = false;
if (options.video_start_bitrate.Get(&options_start_bitrate) &&
options_start_bitrate != send_start_bitrate_) {
send_start_bitrate_ = options_start_bitrate;
start_bitrate_changed = true;
}
bool reset_send_codec_needed = send_codec_ &&
(send_max_bitrate_ != expected_bitrate || denoiser_changed ||
adjusted_min_bitrate || start_bitrate_changed);
if (reset_send_codec_needed) {
// On success, SetSendCodec() will reset send_max_bitrate_ to
// expected_bitrate.
if (!SetSendCodec(*send_codec_,
send_min_bitrate_,
send_start_bitrate_,
expected_bitrate)) {
return false;
}
LogSendCodecChange("SetOptions()");
}
if (leaky_bucket_changed) {
bool enable_leaky_bucket =
options_.video_leaky_bucket.GetWithDefaultIfUnset(false);
LOG(LS_INFO) << "Leaky bucket is enabled : " << enable_leaky_bucket;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
if (engine()->vie()->rtp()->SetTransmissionSmoothingStatus(
it->second->channel_id(), enable_leaky_bucket) != 0) {
LOG_RTCERR2(SetTransmissionSmoothingStatus, it->second->channel_id(),
enable_leaky_bucket);
}
}
}
if (buffer_latency_changed) {
int buffer_latency =
options_.buffered_mode_latency.GetWithDefaultIfUnset(
cricket::kBufferedModeDisabled);
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
if (engine()->vie()->rtp()->SetSenderBufferingMode(
it->second->channel_id(), buffer_latency) != 0) {
LOG_RTCERR2(SetSenderBufferingMode, it->second->channel_id(),
buffer_latency);
}
}
for (RecvChannelMap::iterator it = recv_channels_.begin();
it != recv_channels_.end(); ++it) {
if (engine()->vie()->rtp()->SetReceiverBufferingMode(
it->second->channel_id(), buffer_latency) != 0) {
LOG_RTCERR2(SetReceiverBufferingMode, it->second->channel_id(),
buffer_latency);
}
}
}
if (cpu_overuse_detection_changed) {
bool cpu_overuse_detection =
options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
send_channel->SetCpuOveruseDetection(cpu_overuse_detection);
}
}
if (dscp_option_changed) {
talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
dscp = kVideoDscpValue;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for video channel";
}
}
if (suspend_below_min_bitrate_changed) {
if (options_.suspend_below_min_bitrate.GetWithDefaultIfUnset(false)) {
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->codec()->SuspendBelowMinBitrate(
it->second->channel_id());
}
} else {
LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
}
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (improved_wifi_bwe_changed) {
webrtc::Config config;
config.Set(new webrtc::AimdRemoteRateControl(
options_.use_improved_wifi_bandwidth_estimator
.GetWithDefaultIfUnset(false)));
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->network()->SetBandwidthEstimationConfig(
it->second->channel_id(), config);
}
}
webrtc::CpuOveruseOptions overuse_options;
if (GetCpuOveruseOptions(options_, &overuse_options)) {
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
if (engine()->vie()->base()->SetCpuOveruseOptions(
it->second->channel_id(), overuse_options) != 0) {
LOG_RTCERR1(SetCpuOveruseOptions, it->second->channel_id());
}
}
}
#endif
return true;
}
void WebRtcVideoMediaChannel::SetInterface(NetworkInterface* iface) {
MediaChannel::SetInterface(iface);
// Set the RTP recv/send buffer to a bigger size
MediaChannel::SetOption(NetworkInterface::ST_RTP,
talk_base::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
// TODO(sriniv): Remove or re-enable this.
// As part of b/8030474, send-buffer is size now controlled through
// portallocator flags.
// network_interface_->SetOption(NetworkInterface::ST_RTP,
// talk_base::Socket::OPT_SNDBUF,
// kVideoRtpBufferSize);
}
void WebRtcVideoMediaChannel::UpdateAspectRatio(int ratio_w, int ratio_h) {
ASSERT(ratio_w != 0);
ASSERT(ratio_h != 0);
ratio_w_ = ratio_w;
ratio_h_ = ratio_h;
// For now assume that all streams want the same aspect ratio.
// TODO(hellner): remove the need for this assumption.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
VideoCapturer* capturer = send_channel->video_capturer();
if (capturer) {
capturer->UpdateAspectRatio(ratio_w, ratio_h);
}
}
}
bool WebRtcVideoMediaChannel::GetRenderer(uint32 ssrc,
VideoRenderer** renderer) {
RecvChannelMap::const_iterator it = recv_channels_.find(ssrc);
if (it == recv_channels_.end()) {
if (first_receive_ssrc_ == ssrc &&
recv_channels_.find(0) != recv_channels_.end()) {
LOG(LS_INFO) << " GetRenderer " << ssrc
<< " reuse default renderer #"
<< vie_channel_;
*renderer = recv_channels_[0]->render_adapter()->renderer();
return true;
}
return false;
}
*renderer = it->second->render_adapter()->renderer();
return true;
}
bool WebRtcVideoMediaChannel::GetVideoAdapter(
uint32 ssrc, CoordinatedVideoAdapter** video_adapter) {
SendChannelMap::iterator it = send_channels_.find(ssrc);
if (it == send_channels_.end()) {
return false;
}
*video_adapter = it->second->video_adapter();
return true;
}
void WebRtcVideoMediaChannel::SendFrame(VideoCapturer* capturer,
const VideoFrame* frame) {
// If the |capturer| is registered to any send channel, then send the frame
// to those send channels.
bool capturer_is_channel_owned = false;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == capturer) {
SendFrame(send_channel, frame, capturer->IsScreencast());
capturer_is_channel_owned = true;
}
}
if (capturer_is_channel_owned) {
return;
}
// TODO(hellner): Remove below for loop once the captured frame no longer
// come from the engine, i.e. the engine no longer owns a capturer.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_channel->video_capturer() == NULL) {
SendFrame(send_channel, frame, capturer->IsScreencast());
}
}
}
bool WebRtcVideoMediaChannel::SendFrame(
WebRtcVideoChannelSendInfo* send_channel,
const VideoFrame* frame,
bool is_screencast) {
if (!send_channel) {
return false;
}
if (!send_codec_) {
// Send codec has not been set. No reason to process the frame any further.
return false;
}
const VideoFormat& video_format = send_channel->video_format();
// If the frame should be dropped.
const bool video_format_set = video_format != cricket::VideoFormat();
if (video_format_set &&
(video_format.width == 0 && video_format.height == 0)) {
return true;
}
// Checks if we need to reset vie send codec.
if (!MaybeResetVieSendCodec(send_channel,
static_cast<int>(frame->GetWidth()),
static_cast<int>(frame->GetHeight()),
is_screencast, NULL)) {
LOG(LS_ERROR) << "MaybeResetVieSendCodec failed with "
<< frame->GetWidth() << "x" << frame->GetHeight();
return false;
}
const VideoFrame* frame_out = frame;
talk_base::scoped_ptr<VideoFrame> processed_frame;
// Disable muting for screencast.
const bool mute = (send_channel->muted() && !is_screencast);
send_channel->ProcessFrame(*frame_out, mute, processed_frame.use());
if (processed_frame) {
frame_out = processed_frame.get();
}
webrtc::ViEVideoFrameI420 frame_i420;
// TODO(ronghuawu): Update the webrtc::ViEVideoFrameI420
// to use const unsigned char*
frame_i420.y_plane = const_cast<unsigned char*>(frame_out->GetYPlane());
frame_i420.u_plane = const_cast<unsigned char*>(frame_out->GetUPlane());
frame_i420.v_plane = const_cast<unsigned char*>(frame_out->GetVPlane());
frame_i420.y_pitch = frame_out->GetYPitch();
frame_i420.u_pitch = frame_out->GetUPitch();
frame_i420.v_pitch = frame_out->GetVPitch();
frame_i420.width = static_cast<uint16>(frame_out->GetWidth());
frame_i420.height = static_cast<uint16>(frame_out->GetHeight());
int64 timestamp_ntp_ms = 0;
// TODO(justinlin): Reenable after Windows issues with clock drift are fixed.
// Currently reverted to old behavior of discarding capture timestamp.
#if 0
static const int kTimestampDeltaInSecondsForWarning = 2;
// If the frame timestamp is 0, we will use the deliver time.
const int64 frame_timestamp = frame->GetTimeStamp();
if (frame_timestamp != 0) {
if (abs(time(NULL) - frame_timestamp / talk_base::kNumNanosecsPerSec) >
kTimestampDeltaInSecondsForWarning) {
LOG(LS_WARNING) << "Frame timestamp differs by more than "
<< kTimestampDeltaInSecondsForWarning << " seconds from "
<< "current Unix timestamp.";
}
timestamp_ntp_ms =
talk_base::UnixTimestampNanosecsToNtpMillisecs(frame_timestamp);
}
#endif
return send_channel->external_capture()->IncomingFrameI420(
frame_i420, timestamp_ntp_ms) == 0;
}
bool WebRtcVideoMediaChannel::CreateChannel(uint32 ssrc_key,
MediaDirection direction,
int* channel_id) {
// There are 3 types of channels. Sending only, receiving only and
// sending and receiving. The sending and receiving channel is the
// default channel and there is only one. All other channels that are created
// are associated with the default channel which must exist. The default
// channel id is stored in |vie_channel_|. All channels need to know about
// the default channel to properly handle remb which is why there are
// different ViE create channel calls.
// For this channel the local and remote ssrc key is 0. However, it may
// have a non-zero local and/or remote ssrc depending on if it is currently
// sending and/or receiving.
if ((vie_channel_ == -1 || direction == MD_SENDRECV) &&
(!send_channels_.empty() || !recv_channels_.empty())) {
ASSERT(false);
return false;
}
*channel_id = -1;
if (direction == MD_RECV) {
// All rec channels are associated with the default channel |vie_channel_|
if (engine_->vie()->base()->CreateReceiveChannel(*channel_id,
vie_channel_) != 0) {
LOG_RTCERR2(CreateReceiveChannel, *channel_id, vie_channel_);
return false;
}
} else if (direction == MD_SEND) {
if (engine_->vie()->base()->CreateChannel(*channel_id,
vie_channel_) != 0) {
LOG_RTCERR2(CreateChannel, *channel_id, vie_channel_);
return false;
}
} else {
ASSERT(direction == MD_SENDRECV);
if (engine_->vie()->base()->CreateChannel(*channel_id) != 0) {
LOG_RTCERR1(CreateChannel, *channel_id);
return false;
}
}
if (!ConfigureChannel(*channel_id, direction, ssrc_key)) {
engine_->vie()->base()->DeleteChannel(*channel_id);
*channel_id = -1;
return false;
}
return true;
}
bool WebRtcVideoMediaChannel::CreateUnsignalledRecvChannel(
uint32 ssrc_key, int* out_channel_id) {
int unsignalled_recv_channel_limit =
options_.unsignalled_recv_stream_limit.GetWithDefaultIfUnset(
kNumDefaultUnsignalledVideoRecvStreams);
if (num_unsignalled_recv_channels_ >= unsignalled_recv_channel_limit) {
return false;
}
if (!CreateChannel(ssrc_key, MD_RECV, out_channel_id)) {
return false;
}
// TODO(tvsriram): Support dynamic sizing of unsignalled recv channels.
num_unsignalled_recv_channels_++;
return true;
}
bool WebRtcVideoMediaChannel::ConfigureChannel(int channel_id,
MediaDirection direction,
uint32 ssrc_key) {
const bool receiving = (direction == MD_RECV) || (direction == MD_SENDRECV);
const bool sending = (direction == MD_SEND) || (direction == MD_SENDRECV);
// Register external transport.
if (engine_->vie()->network()->RegisterSendTransport(
channel_id, *this) != 0) {
LOG_RTCERR1(RegisterSendTransport, channel_id);
return false;
}
// Set MTU.
if (engine_->vie()->network()->SetMTU(channel_id, kVideoMtu) != 0) {
LOG_RTCERR2(SetMTU, channel_id, kVideoMtu);
return false;
}
// Turn on RTCP and loss feedback reporting.
if (engine()->vie()->rtp()->SetRTCPStatus(
channel_id, webrtc::kRtcpCompound_RFC4585) != 0) {
LOG_RTCERR2(SetRTCPStatus, channel_id, webrtc::kRtcpCompound_RFC4585);
return false;
}
// Enable pli as key frame request method.
if (engine_->vie()->rtp()->SetKeyFrameRequestMethod(
channel_id, webrtc::kViEKeyFrameRequestPliRtcp) != 0) {
LOG_RTCERR2(SetKeyFrameRequestMethod,
channel_id, webrtc::kViEKeyFrameRequestPliRtcp);
return false;
}
if (!SetNackFec(channel_id, send_red_type_, send_fec_type_, nack_enabled_)) {
// Logged in SetNackFec. Don't spam the logs.
return false;
}
// Note that receiving must always be configured before sending to ensure
// that send and receive channel is configured correctly (ConfigureReceiving
// assumes no sending).
if (receiving) {
if (!ConfigureReceiving(channel_id, ssrc_key)) {
return false;
}
}
if (sending) {
if (!ConfigureSending(channel_id, ssrc_key)) {
return false;
}
}
// Start receiving for both receive and send channels so that we get incoming
// RTP (if receiving) as well as RTCP feedback (if sending).
if (engine()->vie()->base()->StartReceive(channel_id) != 0) {
LOG_RTCERR1(StartReceive, channel_id);
return false;
}
return true;
}
bool WebRtcVideoMediaChannel::ConfigureReceiving(int channel_id,
uint32 remote_ssrc_key) {
// Make sure that an SSRC/key isn't registered more than once.
if (recv_channels_.find(remote_ssrc_key) != recv_channels_.end()) {
return false;
}
// Connect the voice channel, if there is one.
// TODO(perkj): The A/V is synched by the receiving channel. So we need to
// know the SSRC of the remote audio channel in order to fetch the correct
// webrtc VoiceEngine channel. For now- only sync the default channel used
// in 1-1 calls.
if (remote_ssrc_key == 0 && voice_channel_) {
WebRtcVoiceMediaChannel* voice_channel =
static_cast<WebRtcVoiceMediaChannel*>(voice_channel_);
if (engine_->vie()->base()->ConnectAudioChannel(
vie_channel_, voice_channel->voe_channel()) != 0) {
LOG_RTCERR2(ConnectAudioChannel, channel_id,
voice_channel->voe_channel());
LOG(LS_WARNING) << "A/V not synchronized";
// Not a fatal error.
}
}
talk_base::scoped_ptr<WebRtcVideoChannelRecvInfo> channel_info(
new WebRtcVideoChannelRecvInfo(channel_id));
// Install a render adapter.
if (engine_->vie()->render()->AddRenderer(channel_id,
webrtc::kVideoI420, channel_info->render_adapter()) != 0) {
LOG_RTCERR3(AddRenderer, channel_id, webrtc::kVideoI420,
channel_info->render_adapter());
return false;
}
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
kNotSending,
remb_enabled_) != 0) {
LOG_RTCERR3(SetRembStatus, channel_id, kNotSending, remb_enabled_);
return false;
}
if (!SetHeaderExtension(&webrtc::ViERTP_RTCP::SetReceiveTimestampOffsetStatus,
channel_id, receive_extensions_, kRtpTimestampOffsetHeaderExtension)) {
return false;
}
if (!SetHeaderExtension(
&webrtc::ViERTP_RTCP::SetReceiveAbsoluteSendTimeStatus, channel_id,
receive_extensions_, kRtpAbsoluteSenderTimeHeaderExtension)) {
return false;
}
if (remote_ssrc_key != 0) {
// Use the same SSRC as our default channel
// (so the RTCP reports are correct).
unsigned int send_ssrc = 0;
webrtc::ViERTP_RTCP* rtp = engine()->vie()->rtp();
if (rtp->GetLocalSSRC(vie_channel_, send_ssrc) == -1) {
LOG_RTCERR2(GetLocalSSRC, vie_channel_, send_ssrc);
return false;
}
if (rtp->SetLocalSSRC(channel_id, send_ssrc) == -1) {
LOG_RTCERR2(SetLocalSSRC, channel_id, send_ssrc);
return false;
}
} // Else this is the the default channel and we don't change the SSRC.
// Disable color enhancement since it is a bit too aggressive.
if (engine()->vie()->image()->EnableColorEnhancement(channel_id,
false) != 0) {
LOG_RTCERR1(EnableColorEnhancement, channel_id);
return false;
}
if (!SetReceiveCodecs(channel_info.get())) {
return false;
}
int buffer_latency =
options_.buffered_mode_latency.GetWithDefaultIfUnset(
cricket::kBufferedModeDisabled);
if (buffer_latency != cricket::kBufferedModeDisabled) {
if (engine()->vie()->rtp()->SetReceiverBufferingMode(
channel_id, buffer_latency) != 0) {
LOG_RTCERR2(SetReceiverBufferingMode, channel_id, buffer_latency);
}
}
if (render_started_) {
if (engine_->vie()->render()->StartRender(channel_id) != 0) {
LOG_RTCERR1(StartRender, channel_id);
return false;
}
}
// Register decoder observer for incoming framerate and bitrate.
if (engine()->vie()->codec()->RegisterDecoderObserver(
channel_id, *channel_info->decoder_observer()) != 0) {
LOG_RTCERR1(RegisterDecoderObserver, channel_info->decoder_observer());
return false;
}
recv_channels_[remote_ssrc_key] = channel_info.release();
return true;
}
bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id,
uint32 local_ssrc_key) {
// The ssrc key can be zero or correspond to an SSRC.
// Make sure the default channel isn't configured more than once.
if (local_ssrc_key == 0 && send_channels_.find(0) != send_channels_.end()) {
return false;
}
// Make sure that the SSRC is not already in use.
uint32 dummy_key;
if (GetSendChannelKey(local_ssrc_key, &dummy_key)) {
return false;
}
int vie_capture = 0;
webrtc::ViEExternalCapture* external_capture = NULL;
// Register external capture.
if (engine()->vie()->capture()->AllocateExternalCaptureDevice(
vie_capture, external_capture) != 0) {
LOG_RTCERR0(AllocateExternalCaptureDevice);
return false;
}
// Connect external capture.
if (engine()->vie()->capture()->ConnectCaptureDevice(
vie_capture, channel_id) != 0) {
LOG_RTCERR2(ConnectCaptureDevice, vie_capture, channel_id);
return false;
}
talk_base::scoped_ptr<WebRtcVideoChannelSendInfo> send_channel(
new WebRtcVideoChannelSendInfo(channel_id, vie_capture,
external_capture,
engine()->cpu_monitor()));
send_channel->ApplyCpuOptions(options_);
send_channel->SignalCpuAdaptationUnable.connect(this,
&WebRtcVideoMediaChannel::OnCpuAdaptationUnable);
if (options_.cpu_overuse_detection.GetWithDefaultIfUnset(false)) {
send_channel->SetCpuOveruseDetection(true);
}
#ifdef USE_WEBRTC_DEV_BRANCH
webrtc::CpuOveruseOptions overuse_options;
if (GetCpuOveruseOptions(options_, &overuse_options)) {
if (engine()->vie()->base()->SetCpuOveruseOptions(channel_id,
overuse_options) != 0) {
LOG_RTCERR1(SetCpuOveruseOptions, channel_id);
}
}
#endif
// Register encoder observer for outgoing framerate and bitrate.
if (engine()->vie()->codec()->RegisterEncoderObserver(
channel_id, *send_channel->encoder_observer()) != 0) {
LOG_RTCERR1(RegisterEncoderObserver, send_channel->encoder_observer());
return false;
}
if (!SetHeaderExtension(&webrtc::ViERTP_RTCP::SetSendTimestampOffsetStatus,
channel_id, send_extensions_, kRtpTimestampOffsetHeaderExtension)) {
return false;
}
if (!SetHeaderExtension(&webrtc::ViERTP_RTCP::SetSendAbsoluteSendTimeStatus,
channel_id, send_extensions_, kRtpAbsoluteSenderTimeHeaderExtension)) {
return false;
}
if (options_.video_leaky_bucket.GetWithDefaultIfUnset(false)) {
if (engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
true) != 0) {
LOG_RTCERR2(SetTransmissionSmoothingStatus, channel_id, true);
return false;
}
}
int buffer_latency =
options_.buffered_mode_latency.GetWithDefaultIfUnset(
cricket::kBufferedModeDisabled);
if (buffer_latency != cricket::kBufferedModeDisabled) {
if (engine()->vie()->rtp()->SetSenderBufferingMode(
channel_id, buffer_latency) != 0) {
LOG_RTCERR2(SetSenderBufferingMode, channel_id, buffer_latency);
}
}
if (options_.suspend_below_min_bitrate.GetWithDefaultIfUnset(false)) {
engine()->vie()->codec()->SuspendBelowMinBitrate(channel_id);
}
// The remb status direction correspond to the RTP stream (and not the RTCP
// stream). I.e. if send remb is enabled it means it is receiving remote
// rembs and should use them to estimate bandwidth. Receive remb mean that
// remb packets will be generated and that the channel should be included in
// it. If remb is enabled all channels are allowed to contribute to the remb
// but only receive channels will ever end up actually contributing. This
// keeps the logic simple.
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
remb_enabled_,
remb_enabled_) != 0) {
LOG_RTCERR3(SetRembStatus, channel_id, remb_enabled_, remb_enabled_);
return false;
}
if (!SetNackFec(channel_id, send_red_type_, send_fec_type_, nack_enabled_)) {
// Logged in SetNackFec. Don't spam the logs.
return false;
}
send_channels_[local_ssrc_key] = send_channel.release();
return true;
}
bool WebRtcVideoMediaChannel::SetNackFec(int channel_id,
int red_payload_type,
int fec_payload_type,
bool nack_enabled) {
bool enable = (red_payload_type != -1 && fec_payload_type != -1 &&
!InConferenceMode());
if (enable) {
if (engine_->vie()->rtp()->SetHybridNACKFECStatus(
channel_id, nack_enabled, red_payload_type, fec_payload_type) != 0) {
LOG_RTCERR4(SetHybridNACKFECStatus,
channel_id, nack_enabled, red_payload_type, fec_payload_type);
return false;
}
LOG(LS_INFO) << "Hybrid NACK/FEC enabled for channel " << channel_id;
} else {
if (engine_->vie()->rtp()->SetNACKStatus(channel_id, nack_enabled) != 0) {
LOG_RTCERR1(SetNACKStatus, channel_id);
return false;
}
std::string enabled = nack_enabled ? "enabled" : "disabled";
LOG(LS_INFO) << "NACK " << enabled << " for channel " << channel_id;
}
return true;
}
bool WebRtcVideoMediaChannel::SetSendCodec(const webrtc::VideoCodec& codec,
int min_bitrate,
int start_bitrate,
int max_bitrate) {
bool ret_val = true;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
ret_val = SetSendCodec(send_channel, codec, min_bitrate, start_bitrate,
max_bitrate) && ret_val;
}
if (ret_val) {
// All SetSendCodec calls were successful. Update the global state
// accordingly.
send_codec_.reset(new webrtc::VideoCodec(codec));
send_min_bitrate_ = min_bitrate;
send_start_bitrate_ = start_bitrate;
send_max_bitrate_ = max_bitrate;
} else {
// At least one SetSendCodec call failed, rollback.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
if (send_codec_) {
SetSendCodec(send_channel, *send_codec_.get(), send_min_bitrate_,
send_start_bitrate_, send_max_bitrate_);
}
}
}
return ret_val;
}
bool WebRtcVideoMediaChannel::SetSendCodec(
WebRtcVideoChannelSendInfo* send_channel,
const webrtc::VideoCodec& codec,
int min_bitrate,
int start_bitrate,
int max_bitrate) {
if (!send_channel) {
return false;
}
const int channel_id = send_channel->channel_id();
// Make a copy of the codec
webrtc::VideoCodec target_codec = codec;
target_codec.startBitrate = start_bitrate;
target_codec.minBitrate = min_bitrate;
target_codec.maxBitrate = max_bitrate;
// Set the default number of temporal layers for VP8.
if (webrtc::kVideoCodecVP8 == codec.codecType) {
target_codec.codecSpecific.VP8.numberOfTemporalLayers =
kDefaultNumberOfTemporalLayers;
// Turn off the VP8 error resilience
target_codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
bool enable_denoising =
options_.video_noise_reduction.GetWithDefaultIfUnset(false);
target_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
}
// Register external encoder if codec type is supported by encoder factory.
if (engine()->IsExternalEncoderCodecType(codec.codecType) &&
!send_channel->IsEncoderRegistered(target_codec.plType)) {
webrtc::VideoEncoder* encoder =
engine()->CreateExternalEncoder(codec.codecType);
if (encoder) {
if (engine()->vie()->ext_codec()->RegisterExternalSendCodec(
channel_id, target_codec.plType, encoder, false) == 0) {
send_channel->RegisterEncoder(target_codec.plType, encoder);
} else {
LOG_RTCERR2(RegisterExternalSendCodec, channel_id, target_codec.plName);
engine()->DestroyExternalEncoder(encoder);
}
}
}
// Resolution and framerate may vary for different send channels.
const VideoFormat& video_format = send_channel->video_format();
UpdateVideoCodec(video_format, &target_codec);
if (target_codec.width == 0 && target_codec.height == 0) {
const uint32 ssrc = send_channel->stream_params()->first_ssrc();
LOG(LS_INFO) << "0x0 resolution selected. Captured frames will be dropped "
<< "for ssrc: " << ssrc << ".";
} else {
MaybeChangeStartBitrate(channel_id, &target_codec);
webrtc::VideoCodec current_codec;
if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) {
// Compare against existing configured send codec.
if (current_codec == target_codec) {
// Codec is already configured on channel. no need to apply.
return true;
}
}
if (0 != engine()->vie()->codec()->SetSendCodec(channel_id, target_codec)) {
LOG_RTCERR2(SetSendCodec, channel_id, target_codec.plName);
return false;
}
// NOTE: SetRtxSendPayloadType must be called after all simulcast SSRCs
// are configured. Otherwise ssrc's configured after this point will use
// the primary PT for RTX.
if (send_rtx_type_ != -1 &&
engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
send_rtx_type_) != 0) {
LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
return false;
}
}
send_channel->set_interval(
cricket::VideoFormat::FpsToInterval(target_codec.maxFramerate));
return true;
}
static std::string ToString(webrtc::VideoCodecComplexity complexity) {
switch (complexity) {
case webrtc::kComplexityNormal:
return "normal";
case webrtc::kComplexityHigh:
return "high";
case webrtc::kComplexityHigher:
return "higher";
case webrtc::kComplexityMax:
return "max";
default:
return "unknown";
}
}
static std::string ToString(webrtc::VP8ResilienceMode resilience) {
switch (resilience) {
case webrtc::kResilienceOff:
return "off";
case webrtc::kResilientStream:
return "stream";
case webrtc::kResilientFrames:
return "frames";
default:
return "unknown";
}
}
void WebRtcVideoMediaChannel::LogSendCodecChange(const std::string& reason) {
webrtc::VideoCodec vie_codec;
if (engine()->vie()->codec()->GetSendCodec(vie_channel_, vie_codec) != 0) {
LOG_RTCERR1(GetSendCodec, vie_channel_);
return;
}
LOG(LS_INFO) << reason << " : selected video codec "
<< vie_codec.plName << "/"
<< vie_codec.width << "x" << vie_codec.height << "x"
<< static_cast<int>(vie_codec.maxFramerate) << "fps"
<< "@" << vie_codec.maxBitrate << "kbps"
<< " (min=" << vie_codec.minBitrate << "kbps,"
<< " start=" << vie_codec.startBitrate << "kbps)";
LOG(LS_INFO) << "Video max quantization: " << vie_codec.qpMax;
if (webrtc::kVideoCodecVP8 == vie_codec.codecType) {
LOG(LS_INFO) << "VP8 number of temporal layers: "
<< static_cast<int>(
vie_codec.codecSpecific.VP8.numberOfTemporalLayers);
LOG(LS_INFO) << "VP8 options : "
<< "picture loss indication = "
<< vie_codec.codecSpecific.VP8.pictureLossIndicationOn
<< ", feedback mode = "
<< vie_codec.codecSpecific.VP8.feedbackModeOn
<< ", complexity = "
<< ToString(vie_codec.codecSpecific.VP8.complexity)
<< ", resilience = "
<< ToString(vie_codec.codecSpecific.VP8.resilience)
<< ", denoising = "
<< vie_codec.codecSpecific.VP8.denoisingOn
<< ", error concealment = "
<< vie_codec.codecSpecific.VP8.errorConcealmentOn
<< ", automatic resize = "
<< vie_codec.codecSpecific.VP8.automaticResizeOn
<< ", frame dropping = "
<< vie_codec.codecSpecific.VP8.frameDroppingOn
<< ", key frame interval = "
<< vie_codec.codecSpecific.VP8.keyFrameInterval;
}
if (send_rtx_type_ != -1) {
LOG(LS_INFO) << "RTX payload type: " << send_rtx_type_;
}
}
bool WebRtcVideoMediaChannel::SetReceiveCodecs(
WebRtcVideoChannelRecvInfo* info) {
int red_type = -1;
int fec_type = -1;
int channel_id = info->channel_id();
for (std::vector<webrtc::VideoCodec>::iterator it = receive_codecs_.begin();
it != receive_codecs_.end(); ++it) {
if (it->codecType == webrtc::kVideoCodecRED) {
red_type = it->plType;
} else if (it->codecType == webrtc::kVideoCodecULPFEC) {
fec_type = it->plType;
}
if (engine()->vie()->codec()->SetReceiveCodec(channel_id, *it) != 0) {
LOG_RTCERR2(SetReceiveCodec, channel_id, it->plName);
return false;
}
if (!info->IsDecoderRegistered(it->plType) &&
it->codecType != webrtc::kVideoCodecRED &&
it->codecType != webrtc::kVideoCodecULPFEC) {
webrtc::VideoDecoder* decoder =
engine()->CreateExternalDecoder(it->codecType);
if (decoder) {
if (engine()->vie()->ext_codec()->RegisterExternalReceiveCodec(
channel_id, it->plType, decoder) == 0) {
info->RegisterDecoder(it->plType, decoder);
} else {
LOG_RTCERR2(RegisterExternalReceiveCodec, channel_id, it->plName);
engine()->DestroyExternalDecoder(decoder);
}
}
}
}
return true;
}
int WebRtcVideoMediaChannel::GetRecvChannelNum(uint32 ssrc) {
if (ssrc == first_receive_ssrc_) {
return vie_channel_;
}
RecvChannelMap::iterator it = recv_channels_.find(ssrc);
return (it != recv_channels_.end()) ? it->second->channel_id() : -1;
}
// If the new frame size is different from the send codec size we set on vie,
// we need to reset the send codec on vie.
// The new send codec size should not exceed send_codec_ which is controlled
// only by the 'jec' logic.
bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
WebRtcVideoChannelSendInfo* send_channel,
int new_width,
int new_height,
bool is_screencast,
bool* reset) {
if (reset) {
*reset = false;
}
ASSERT(send_codec_.get() != NULL);
webrtc::VideoCodec target_codec = *send_codec_.get();
const VideoFormat& video_format = send_channel->video_format();
UpdateVideoCodec(video_format, &target_codec);
// Vie send codec size should not exceed target_codec.
int target_width = new_width;
int target_height = new_height;
if (!is_screencast &&
(new_width > target_codec.width || new_height > target_codec.height)) {
target_width = target_codec.width;
target_height = target_codec.height;
}
// Get current vie codec.
webrtc::VideoCodec vie_codec;
const int channel_id = send_channel->channel_id();
if (engine()->vie()->codec()->GetSendCodec(channel_id, vie_codec) != 0) {
LOG_RTCERR1(GetSendCodec, channel_id);
return false;
}
const int cur_width = vie_codec.width;
const int cur_height = vie_codec.height;
// Only reset send codec when there is a size change. Additionally,
// automatic resize needs to be turned off when screencasting and on when
// not screencasting.
// Don't allow automatic resizing for screencasting.
bool automatic_resize = !is_screencast;
// Turn off VP8 frame dropping when screensharing as the current model does
// not work well at low fps.
bool vp8_frame_dropping = !is_screencast;
// Disable denoising for screencasting.
bool enable_denoising =
options_.video_noise_reduction.GetWithDefaultIfUnset(false);
#ifdef USE_WEBRTC_DEV_BRANCH
int screencast_min_bitrate =
options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
bool leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(false);
#endif
bool denoising = !is_screencast && enable_denoising;
bool reset_send_codec =
target_width != cur_width || target_height != cur_height ||
automatic_resize != vie_codec.codecSpecific.VP8.automaticResizeOn ||
denoising != vie_codec.codecSpecific.VP8.denoisingOn ||
vp8_frame_dropping != vie_codec.codecSpecific.VP8.frameDroppingOn;
if (reset_send_codec) {
// Set the new codec on vie.
vie_codec.width = target_width;
vie_codec.height = target_height;
vie_codec.maxFramerate = target_codec.maxFramerate;
vie_codec.startBitrate = target_codec.startBitrate;
#ifdef USE_WEBRTC_DEV_BRANCH
vie_codec.targetBitrate = 0;
#endif
vie_codec.codecSpecific.VP8.automaticResizeOn = automatic_resize;
vie_codec.codecSpecific.VP8.denoisingOn = denoising;
vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
bool maybe_change_start_bitrate = !is_screencast;
#ifdef USE_WEBRTC_DEV_BRANCH
// TODO(pbos): When USE_WEBRTC_DEV_BRANCH is removed, remove
// maybe_change_start_bitrate as well. MaybeChangeStartBitrate should be
// called for all content.
maybe_change_start_bitrate = true;
#endif
if (maybe_change_start_bitrate)
MaybeChangeStartBitrate(channel_id, &vie_codec);
if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) {
LOG_RTCERR1(SetSendCodec, channel_id);
return false;
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (is_screencast) {
engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
screencast_min_bitrate);
// If screencast and min bitrate set, force enable pacer.
if (screencast_min_bitrate > 0) {
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
true);
}
} else {
// In case of switching from screencast to regular capture, set
// min bitrate padding and pacer back to defaults.
engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0);
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
leaky_bucket);
}
#endif
if (reset) {
*reset = true;
}
LogSendCodecChange("Capture size changed");
}
return true;
}
void WebRtcVideoMediaChannel::MaybeChangeStartBitrate(
int channel_id, webrtc::VideoCodec* video_codec) {
if (video_codec->startBitrate < video_codec->minBitrate) {
video_codec->startBitrate = video_codec->minBitrate;
} else if (video_codec->startBitrate > video_codec->maxBitrate) {
video_codec->startBitrate = video_codec->maxBitrate;
}
// Use a previous target bitrate, if there is one.
unsigned int current_target_bitrate = 0;
if (engine()->vie()->codec()->GetCodecTargetBitrate(
channel_id, &current_target_bitrate) == 0) {
// Convert to kbps.
current_target_bitrate /= 1000;
if (current_target_bitrate > video_codec->maxBitrate) {
current_target_bitrate = video_codec->maxBitrate;
}
if (current_target_bitrate > video_codec->startBitrate) {
video_codec->startBitrate = current_target_bitrate;
}
}
}
void WebRtcVideoMediaChannel::OnMessage(talk_base::Message* msg) {
FlushBlackFrameData* black_frame_data =
static_cast<FlushBlackFrameData*>(msg->pdata);
FlushBlackFrame(black_frame_data->ssrc, black_frame_data->timestamp);
delete black_frame_data;
}
int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data,
int len) {
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendPacket(&packet) ? len : -1;
}
int WebRtcVideoMediaChannel::SendRTCPPacket(int channel,
const void* data,
int len) {
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
return MediaChannel::SendRtcp(&packet) ? len : -1;
}
void WebRtcVideoMediaChannel::QueueBlackFrame(uint32 ssrc, int64 timestamp,
int framerate) {
if (timestamp) {
FlushBlackFrameData* black_frame_data = new FlushBlackFrameData(
ssrc,
timestamp);
const int delay_ms = static_cast<int>(
2 * cricket::VideoFormat::FpsToInterval(framerate) *
talk_base::kNumMillisecsPerSec / talk_base::kNumNanosecsPerSec);
worker_thread()->PostDelayed(delay_ms, this, 0, black_frame_data);
}
}
void WebRtcVideoMediaChannel::FlushBlackFrame(uint32 ssrc, int64 timestamp) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannel(ssrc);
if (!send_channel) {
return;
}
talk_base::scoped_ptr<const VideoFrame> black_frame_ptr;
const WebRtcLocalStreamInfo* channel_stream_info =
send_channel->local_stream_info();
int64 last_frame_time_stamp = channel_stream_info->time_stamp();
if (last_frame_time_stamp == timestamp) {
size_t last_frame_width = 0;
size_t last_frame_height = 0;
int64 last_frame_elapsed_time = 0;
channel_stream_info->GetLastFrameInfo(&last_frame_width, &last_frame_height,
&last_frame_elapsed_time);
if (!last_frame_width || !last_frame_height) {
return;
}
WebRtcVideoFrame black_frame;
// Black frame is not screencast.
const bool screencasting = false;
const int64 timestamp_delta = send_channel->interval();
if (!black_frame.InitToBlack(send_codec_->width, send_codec_->height, 1, 1,
last_frame_elapsed_time + timestamp_delta,
last_frame_time_stamp + timestamp_delta) ||
!SendFrame(send_channel, &black_frame, screencasting)) {
LOG(LS_ERROR) << "Failed to send black frame.";
}
}
}
void WebRtcVideoMediaChannel::OnCpuAdaptationUnable() {
// ssrc is hardcoded to 0. This message is based on a system wide issue,
// so finding which ssrc caused it doesn't matter.
SignalMediaError(0, VideoMediaChannel::ERROR_REC_CPU_MAX_CANT_DOWNGRADE);
}
void WebRtcVideoMediaChannel::SetNetworkTransmissionState(
bool is_transmitting) {
LOG(LS_INFO) << "SetNetworkTransmissionState: " << is_transmitting;
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
int channel_id = send_channel->channel_id();
engine_->vie()->network()->SetNetworkTransmissionState(channel_id,
is_transmitting);
}
}
bool WebRtcVideoMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
int channel_id, const RtpHeaderExtension* extension) {
bool enable = false;
int id = 0;
if (extension) {
enable = true;
id = extension->id;
}
if ((engine_->vie()->rtp()->*setter)(channel_id, enable, id) != 0) {
LOG_RTCERR4(*setter, extension->uri, channel_id, enable, id);
return false;
}
return true;
}
bool WebRtcVideoMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
int channel_id, const std::vector<RtpHeaderExtension>& extensions,
const char header_extension_uri[]) {
const RtpHeaderExtension* extension = FindHeaderExtension(extensions,
header_extension_uri);
return SetHeaderExtension(setter, channel_id, extension);
}
bool WebRtcVideoMediaChannel::SetLocalRtxSsrc(int channel_id,
const StreamParams& send_params,
uint32 primary_ssrc,
int stream_idx) {
uint32 rtx_ssrc = 0;
bool has_rtx = send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc);
if (has_rtx && engine()->vie()->rtp()->SetLocalSSRC(
channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, stream_idx) != 0) {
LOG_RTCERR4(SetLocalSSRC, channel_id, rtx_ssrc,
webrtc::kViEStreamTypeRtx, stream_idx);
return false;
}
return true;
}
void WebRtcVideoMediaChannel::MaybeConnectCapturer(VideoCapturer* capturer) {
if (capturer != NULL && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.connect(this,
&WebRtcVideoMediaChannel::SendFrame);
}
}
void WebRtcVideoMediaChannel::MaybeDisconnectCapturer(VideoCapturer* capturer) {
if (capturer != NULL && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.disconnect(this);
}
}
} // namespace cricket
#endif // HAVE_WEBRTC_VIDEO