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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/rtpsender.h"
#include <vector>
#include "api/mediastreaminterface.h"
#include "pc/localaudiosource.h"
#include "pc/statscollector.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
// This function is only expected to be called on the signalling thread.
int GenerateUniqueId() {
static int g_unique_id = 0;
return ++g_unique_id;
}
// Returns an true if any RtpEncodingParameters member that isn't implemented
// contains a value.
bool UnimplementedRtpEncodingParameterHasValue(
const RtpEncodingParameters& encoding_params) {
if (encoding_params.codec_payload_type.has_value() ||
encoding_params.fec.has_value() || encoding_params.rtx.has_value() ||
encoding_params.dtx.has_value() || encoding_params.ptime.has_value() ||
encoding_params.max_framerate.has_value() ||
!encoding_params.rid.empty() ||
encoding_params.scale_resolution_down_by.has_value() ||
encoding_params.scale_framerate_down_by.has_value() ||
!encoding_params.dependency_rids.empty()) {
return true;
}
return false;
}
// Returns true if a "per-sender" encoding parameter contains a value that isn't
// its default. Currently max_bitrate_bps and bitrate_priority both are
// implemented "per-sender," meaning that these encoding parameters
// are used for the RtpSender as a whole, not for a specific encoding layer.
// This is done by setting these encoding parameters at index 0 of
// RtpParameters.encodings. This function can be used to check if these
// parameters are set at any index other than 0 of RtpParameters.encodings,
// because they are currently unimplemented to be used for a specific encoding
// layer.
bool PerSenderRtpEncodingParameterHasValue(
const RtpEncodingParameters& encoding_params) {
if (encoding_params.max_bitrate_bps.has_value() ||
encoding_params.bitrate_priority != kDefaultBitratePriority) {
return true;
}
return false;
}
// Returns true if any RtpParameters member that isn't implemented contains a
// value.
bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
if (!parameters.mid.empty() ||
parameters.degradation_preference != DegradationPreference::BALANCED) {
return true;
}
for (size_t i = 0; i < parameters.encodings.size(); ++i) {
if (UnimplementedRtpEncodingParameterHasValue(parameters.encodings[i])) {
return true;
}
// Encoding parameters that are per-sender should only contain value at
// index 0.
if (i != 0 &&
PerSenderRtpEncodingParameterHasValue(parameters.encodings[i])) {
return true;
}
}
return false;
}
} // namespace
LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
rtc::CritScope lock(&lock_);
if (sink_)
sink_->OnClose();
}
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {
rtc::CritScope lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames);
}
}
void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
rtc::CritScope lock(&lock_);
RTC_DCHECK(!sink || !sink_);
sink_ = sink;
}
AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
StatsCollector* stats)
: AudioRtpSender(worker_thread, nullptr, {rtc::CreateRandomUuid()}, stats) {
}
AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<AudioTrackInterface> track,
const std::vector<std::string>& stream_ids,
StatsCollector* stats)
: worker_thread_(worker_thread),
id_(track ? track->id() : rtc::CreateRandomUuid()),
stream_ids_(stream_ids),
stats_(stats),
track_(track),
dtmf_sender_proxy_(DtmfSenderProxy::Create(
rtc::Thread::Current(),
DtmfSender::Create(track_, rtc::Thread::Current(), this))),
cached_track_enabled_(track ? track->enabled() : false),
sink_adapter_(new LocalAudioSinkAdapter()),
attachment_id_(track ? GenerateUniqueId() : 0) {
RTC_DCHECK(worker_thread);
if (track_) {
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
}
}
AudioRtpSender::~AudioRtpSender() {
// For DtmfSender.
SignalDestroyed();
Stop();
}
bool AudioRtpSender::CanInsertDtmf() {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
return false;
}
// Check that this RTP sender is active (description has been applied that
// matches an SSRC to its ID).
if (!ssrc_) {
RTC_LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
return false;
}
return worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&] { return media_channel_->CanInsertDtmf(); });
}
bool AudioRtpSender::InsertDtmf(int code, int duration) {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
return false;
}
if (!ssrc_) {
RTC_LOG(LS_ERROR) << "InsertDtmf: Sender does not have SSRC.";
return false;
}
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->InsertDtmf(ssrc_, code, duration);
});
if (!success) {
RTC_LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
}
return success;
}
sigslot::signal0<>* AudioRtpSender::GetOnDestroyedSignal() {
return &SignalDestroyed;
}
void AudioRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
if (can_send_track()) {
SetAudioSend();
}
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack");
if (stopped_) {
RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
}
if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
RTC_LOG(LS_ERROR) << "SetTrack called on audio RtpSender with "
<< track->kind() << " track.";
return false;
}
AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
// Detach from old track.
if (track_) {
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
}
if (can_send_track() && stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
// Attach to new track.
bool prev_can_send_track = can_send_track();
// Keep a reference to the old track to keep it alive until we call
// SetAudioSend.
rtc::scoped_refptr<AudioTrackInterface> old_track = track_;
track_ = audio_track;
if (track_) {
cached_track_enabled_ = track_->enabled();
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
}
// Update audio channel.
if (can_send_track()) {
SetAudioSend();
if (stats_) {
stats_->AddLocalAudioTrack(track_.get(), ssrc_);
}
} else if (prev_can_send_track) {
ClearAudioSend();
}
attachment_id_ = GenerateUniqueId();
return true;
}
RtpParameters AudioRtpSender::GetParameters() {
if (!media_channel_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
last_transaction_id_ = rtc::CreateRandomUuid();
result.transaction_id = last_transaction_id_.value();
return result;
});
}
RTCError AudioRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
if (!media_channel_ || stopped_) {
return RTCError(RTCErrorType::INVALID_STATE);
}
if (!last_transaction_id_) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
"Failed to set parameters since getParameters() has never been called"
" on this sender");
}
if (last_transaction_id_ != parameters.transaction_id) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Failed to set parameters since the transaction_id doesn't match"
" the last value returned from getParameters()");
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
RTCError result = media_channel_->SetRtpSendParameters(ssrc_, parameters);
last_transaction_id_.reset();
return result;
});
}
rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
return dtmf_sender_proxy_;
}
void AudioRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
// If we are already sending with a particular SSRC, stop sending.
if (can_send_track()) {
ClearAudioSend();
if (stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
}
ssrc_ = ssrc;
if (can_send_track()) {
SetAudioSend();
if (stats_) {
stats_->AddLocalAudioTrack(track_.get(), ssrc_);
}
}
}
void AudioRtpSender::Stop() {
TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
}
if (track_) {
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
}
if (can_send_track()) {
ClearAudioSend();
if (stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
}
media_channel_ = nullptr;
stopped_ = true;
}
void AudioRtpSender::SetAudioSend() {
RTC_DCHECK(!stopped_);
RTC_DCHECK(can_send_track());
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
return;
}
cricket::AudioOptions options;
#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_WEBKIT_BUILD)
// TODO(tommi): Remove this hack when we move CreateAudioSource out of
// PeerConnection. This is a bit of a strange way to apply local audio
// options since it is also applied to all streams/channels, local or remote.
if (track_->enabled() && track_->GetSource() &&
!track_->GetSource()->remote()) {
// TODO(xians): Remove this static_cast since we should be able to connect
// a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
#endif
// |track_->enabled()| hops to the signaling thread, so call it before we hop
// to the worker thread or else it will deadlock.
bool track_enabled = track_->enabled();
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetAudioSend(ssrc_, track_enabled, &options,
sink_adapter_.get());
});
if (!success) {
RTC_LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc_;
}
}
void AudioRtpSender::ClearAudioSend() {
RTC_DCHECK(ssrc_ != 0);
RTC_DCHECK(!stopped_);
if (!media_channel_) {
RTC_LOG(LS_WARNING) << "ClearAudioSend: No audio channel exists.";
return;
}
cricket::AudioOptions options;
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetAudioSend(ssrc_, false, &options, nullptr);
});
if (!success) {
RTC_LOG(LS_WARNING) << "ClearAudioSend: ssrc is incorrect: " << ssrc_;
}
}
VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread)
: VideoRtpSender(worker_thread, nullptr, {rtc::CreateRandomUuid()}) {}
VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread,
rtc::scoped_refptr<VideoTrackInterface> track,
const std::vector<std::string>& stream_ids)
: worker_thread_(worker_thread),
id_(track ? track->id() : rtc::CreateRandomUuid()),
stream_ids_(stream_ids),
track_(track),
cached_track_content_hint_(track
? track->content_hint()
: VideoTrackInterface::ContentHint::kNone),
attachment_id_(track ? GenerateUniqueId() : 0) {
RTC_DCHECK(worker_thread);
if (track_) {
track_->RegisterObserver(this);
}
}
VideoRtpSender::~VideoRtpSender() {
Stop();
}
void VideoRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_content_hint_ != track_->content_hint()) {
cached_track_content_hint_ = track_->content_hint();
if (can_send_track()) {
SetVideoSend();
}
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack");
if (stopped_) {
RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
}
if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
RTC_LOG(LS_ERROR) << "SetTrack called on video RtpSender with "
<< track->kind() << " track.";
return false;
}
VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
// Detach from old track.
if (track_) {
track_->UnregisterObserver(this);
}
// Attach to new track.
bool prev_can_send_track = can_send_track();
// Keep a reference to the old track to keep it alive until we call
// SetVideoSend.
rtc::scoped_refptr<VideoTrackInterface> old_track = track_;
track_ = video_track;
if (track_) {
cached_track_content_hint_ = track_->content_hint();
track_->RegisterObserver(this);
}
// Update video channel.
if (can_send_track()) {
SetVideoSend();
} else if (prev_can_send_track) {
ClearVideoSend();
}
attachment_id_ = GenerateUniqueId();
return true;
}
RtpParameters VideoRtpSender::GetParameters() {
if (!media_channel_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
last_transaction_id_ = rtc::CreateRandomUuid();
result.transaction_id = last_transaction_id_.value();
return result;
});
}
RTCError VideoRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
if (!media_channel_ || stopped_) {
return RTCError(RTCErrorType::INVALID_STATE);
}
if (!last_transaction_id_) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
"Failed to set parameters since getParameters() has never been called"
" on this sender");
}
if (last_transaction_id_ != parameters.transaction_id) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Failed to set parameters since the transaction_id doesn't match"
" the last value returned from getParameters()");
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
RTCError result = media_channel_->SetRtpSendParameters(ssrc_, parameters);
last_transaction_id_.reset();
return result;
});
}
rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
RTC_LOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
return nullptr;
}
void VideoRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
// If we are already sending with a particular SSRC, stop sending.
if (can_send_track()) {
ClearVideoSend();
}
ssrc_ = ssrc;
if (can_send_track()) {
SetVideoSend();
}
}
void VideoRtpSender::Stop() {
TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
}
if (track_) {
track_->UnregisterObserver(this);
}
if (can_send_track()) {
ClearVideoSend();
}
media_channel_ = nullptr;
stopped_ = true;
}
void VideoRtpSender::SetVideoSend() {
RTC_DCHECK(!stopped_);
RTC_DCHECK(can_send_track());
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "SetVideoSend: No video channel exists.";
return;
}
cricket::VideoOptions options;
VideoTrackSourceInterface* source = track_->GetSource();
if (source) {
options.is_screencast = source->is_screencast();
options.video_noise_reduction = source->needs_denoising();
}
switch (cached_track_content_hint_) {
case VideoTrackInterface::ContentHint::kNone:
break;
case VideoTrackInterface::ContentHint::kFluid:
options.is_screencast = false;
break;
case VideoTrackInterface::ContentHint::kDetailed:
options.is_screencast = true;
break;
}
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetVideoSend(ssrc_, &options, track_);
});
RTC_DCHECK(success);
}
void VideoRtpSender::ClearVideoSend() {
RTC_DCHECK(ssrc_ != 0);
RTC_DCHECK(!stopped_);
if (!media_channel_) {
RTC_LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
return;
}
// Allow SetVideoSend to fail since |enable| is false and |source| is null.
// This the normal case when the underlying media channel has already been
// deleted.
worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetVideoSend(ssrc_, nullptr, nullptr);
});
}
} // namespace webrtc