blob: 58fef4041ecfcef3a76fcfb296c923355f68625c [file] [log] [blame]
/*
* 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/rtp_sender.h"
#include <algorithm>
#include <atomic>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "api/audio_options.h"
#include "api/media_stream_interface.h"
#include "api/priority.h"
#include "media/base/media_engine.h"
#include "pc/stats_collector_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
// This function is only expected to be called on the signaling thread.
// On the other hand, some test or even production setups may use
// several signaling threads.
int GenerateUniqueId() {
static std::atomic<int> g_unique_id{0};
return ++g_unique_id;
}
// 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.bitrate_priority != kDefaultBitratePriority ||
encoding_params.network_priority != Priority::kLow) {
return true;
}
return false;
}
void RemoveEncodingLayers(const std::vector<std::string>& rids,
std::vector<RtpEncodingParameters>* encodings) {
RTC_DCHECK(encodings);
encodings->erase(
std::remove_if(encodings->begin(), encodings->end(),
[&rids](const RtpEncodingParameters& encoding) {
return absl::c_linear_search(rids, encoding.rid);
}),
encodings->end());
}
RtpParameters RestoreEncodingLayers(
const RtpParameters& parameters,
const std::vector<std::string>& removed_rids,
const std::vector<RtpEncodingParameters>& all_layers) {
RTC_DCHECK_EQ(parameters.encodings.size() + removed_rids.size(),
all_layers.size());
RtpParameters result(parameters);
result.encodings.clear();
size_t index = 0;
for (const RtpEncodingParameters& encoding : all_layers) {
if (absl::c_linear_search(removed_rids, encoding.rid)) {
result.encodings.push_back(encoding);
continue;
}
result.encodings.push_back(parameters.encodings[index++]);
}
return result;
}
} // namespace
// Returns true if any RtpParameters member that isn't implemented contains a
// value.
bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
if (!parameters.mid.empty()) {
return true;
}
for (size_t i = 0; i < parameters.encodings.size(); ++i) {
// Encoding parameters that are per-sender should only contain value at
// index 0.
if (i != 0 &&
PerSenderRtpEncodingParameterHasValue(parameters.encodings[i])) {
return true;
}
}
return false;
}
RtpSenderBase::RtpSenderBase(rtc::Thread* worker_thread,
const std::string& id,
SetStreamsObserver* set_streams_observer)
: signaling_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
id_(id),
set_streams_observer_(set_streams_observer) {
RTC_DCHECK(worker_thread);
init_parameters_.encodings.emplace_back();
}
void RtpSenderBase::SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
RTC_DCHECK_RUN_ON(signaling_thread_);
frame_encryptor_ = std::move(frame_encryptor);
// Special Case: Set the frame encryptor to any value on any existing channel.
if (media_channel_ && ssrc_ && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_);
});
}
}
void RtpSenderBase::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
media_channel_ = media_channel;
}
RtpParameters RtpSenderBase::GetParametersInternal() const {
RTC_DCHECK_RUN_ON(signaling_thread_);
if (stopped_) {
return RtpParameters();
}
if (!media_channel_ || !ssrc_) {
return init_parameters_;
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
RemoveEncodingLayers(disabled_rids_, &result.encodings);
return result;
});
}
RtpParameters RtpSenderBase::GetParameters() const {
RTC_DCHECK_RUN_ON(signaling_thread_);
RtpParameters result = GetParametersInternal();
last_transaction_id_ = rtc::CreateRandomUuid();
result.transaction_id = last_transaction_id_.value();
return result;
}
RTCError RtpSenderBase::SetParametersInternal(const RtpParameters& parameters) {
RTC_DCHECK_RUN_ON(signaling_thread_);
RTC_DCHECK(!stopped_);
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
if (!media_channel_ || !ssrc_) {
auto result = cricket::CheckRtpParametersInvalidModificationAndValues(
init_parameters_, parameters);
if (result.ok()) {
init_parameters_ = parameters;
}
return result;
}
return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
RtpParameters rtp_parameters = parameters;
if (!disabled_rids_.empty()) {
// Need to add the inactive layers.
RtpParameters old_parameters =
media_channel_->GetRtpSendParameters(ssrc_);
rtp_parameters = RestoreEncodingLayers(parameters, disabled_rids_,
old_parameters.encodings);
}
return media_channel_->SetRtpSendParameters(ssrc_, rtp_parameters);
});
}
RTCError RtpSenderBase::SetParameters(const RtpParameters& parameters) {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "RtpSenderBase::SetParameters");
if (is_transceiver_stopped_) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
"Cannot set parameters on sender of a stopped transceiver.");
}
if (stopped_) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Cannot set parameters on a stopped sender.");
}
if (stopped_) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Cannot set parameters on a stopped sender.");
}
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()");
}
RTCError result = SetParametersInternal(parameters);
last_transaction_id_.reset();
return result;
}
void RtpSenderBase::SetStreams(const std::vector<std::string>& stream_ids) {
set_stream_ids(stream_ids);
if (set_streams_observer_)
set_streams_observer_->OnSetStreams();
}
bool RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "RtpSenderBase::SetTrack");
if (stopped_) {
RTC_LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
}
if (track && track->kind() != track_kind()) {
RTC_LOG(LS_ERROR) << "SetTrack with " << track->kind()
<< " called on RtpSender with " << track_kind()
<< " track.";
return false;
}
// Detach from old track.
if (track_) {
DetachTrack();
track_->UnregisterObserver(this);
RemoveTrackFromStats();
}
// 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 SetSend.
rtc::scoped_refptr<MediaStreamTrackInterface> old_track = track_;
track_ = track;
if (track_) {
track_->RegisterObserver(this);
AttachTrack();
}
// Update channel.
if (can_send_track()) {
SetSend();
AddTrackToStats();
} else if (prev_can_send_track) {
ClearSend();
}
attachment_id_ = (track_ ? GenerateUniqueId() : 0);
return true;
}
void RtpSenderBase::SetSsrc(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "RtpSenderBase::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
// If we are already sending with a particular SSRC, stop sending.
if (can_send_track()) {
ClearSend();
RemoveTrackFromStats();
}
ssrc_ = ssrc;
if (can_send_track()) {
SetSend();
AddTrackToStats();
}
if (!init_parameters_.encodings.empty()) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK(media_channel_);
// Get the current parameters, which are constructed from the SDP.
// The number of layers in the SDP is currently authoritative to support
// SDP munging for Plan-B simulcast with "a=ssrc-group:SIM <ssrc-id>..."
// lines as described in RFC 5576.
// All fields should be default constructed and the SSRC field set, which
// we need to copy.
RtpParameters current_parameters =
media_channel_->GetRtpSendParameters(ssrc_);
RTC_CHECK_GE(current_parameters.encodings.size(),
init_parameters_.encodings.size());
for (size_t i = 0; i < init_parameters_.encodings.size(); ++i) {
init_parameters_.encodings[i].ssrc =
current_parameters.encodings[i].ssrc;
init_parameters_.encodings[i].rid = current_parameters.encodings[i].rid;
current_parameters.encodings[i] = init_parameters_.encodings[i];
}
current_parameters.degradation_preference =
init_parameters_.degradation_preference;
media_channel_->SetRtpSendParameters(ssrc_, current_parameters);
init_parameters_.encodings.clear();
});
}
// Attempt to attach the frame decryptor to the current media channel.
if (frame_encryptor_) {
SetFrameEncryptor(frame_encryptor_);
}
if (frame_transformer_) {
SetEncoderToPacketizerFrameTransformer(frame_transformer_);
}
}
void RtpSenderBase::Stop() {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "RtpSenderBase::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
}
if (track_) {
DetachTrack();
track_->UnregisterObserver(this);
}
if (can_send_track()) {
ClearSend();
RemoveTrackFromStats();
}
media_channel_ = nullptr;
set_streams_observer_ = nullptr;
stopped_ = true;
}
RTCError RtpSenderBase::DisableEncodingLayers(
const std::vector<std::string>& rids) {
RTC_DCHECK_RUN_ON(signaling_thread_);
if (stopped_) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Cannot disable encodings on a stopped sender.");
}
if (rids.empty()) {
return RTCError::OK();
}
// Check that all the specified layers exist and disable them in the channel.
RtpParameters parameters = GetParametersInternal();
for (const std::string& rid : rids) {
if (absl::c_none_of(parameters.encodings,
[&rid](const RtpEncodingParameters& encoding) {
return encoding.rid == rid;
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"RID: " + rid + " does not refer to a valid layer.");
}
}
if (!media_channel_ || !ssrc_) {
RemoveEncodingLayers(rids, &init_parameters_.encodings);
// Invalidate any transaction upon success.
last_transaction_id_.reset();
return RTCError::OK();
}
for (RtpEncodingParameters& encoding : parameters.encodings) {
// Remain active if not in the disable list.
encoding.active &= absl::c_none_of(
rids,
[&encoding](const std::string& rid) { return encoding.rid == rid; });
}
RTCError result = SetParametersInternal(parameters);
if (result.ok()) {
disabled_rids_.insert(disabled_rids_.end(), rids.begin(), rids.end());
// Invalidate any transaction upon success.
last_transaction_id_.reset();
}
return result;
}
void RtpSenderBase::SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(signaling_thread_);
frame_transformer_ = std::move(frame_transformer);
if (media_channel_ && ssrc_ && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel_->SetEncoderToPacketizerFrameTransformer(
ssrc_, frame_transformer_);
});
}
}
LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
MutexLock 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,
absl::optional<int64_t> absolute_capture_timestamp_ms) {
MutexLock lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames, absolute_capture_timestamp_ms);
num_preferred_channels_ = sink_->NumPreferredChannels();
}
}
void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
MutexLock lock(&lock_);
RTC_DCHECK(!sink || !sink_);
sink_ = sink;
}
rtc::scoped_refptr<AudioRtpSender> AudioRtpSender::Create(
rtc::Thread* worker_thread,
const std::string& id,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer) {
return rtc::make_ref_counted<AudioRtpSender>(worker_thread, id, stats,
set_streams_observer);
}
AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
const std::string& id,
StatsCollectorInterface* stats,
SetStreamsObserver* set_streams_observer)
: RtpSenderBase(worker_thread, id, set_streams_observer),
stats_(stats),
dtmf_sender_proxy_(DtmfSenderProxy::Create(
rtc::Thread::Current(),
DtmfSender::Create(rtc::Thread::Current(), this))),
sink_adapter_(new LocalAudioSinkAdapter()) {}
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 voice_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 voice_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() {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
if (can_send_track()) {
SetSend();
}
}
}
void AudioRtpSender::DetachTrack() {
RTC_DCHECK(track_);
audio_track()->RemoveSink(sink_adapter_.get());
}
void AudioRtpSender::AttachTrack() {
RTC_DCHECK(track_);
cached_track_enabled_ = track_->enabled();
audio_track()->AddSink(sink_adapter_.get());
}
void AudioRtpSender::AddTrackToStats() {
if (can_send_track() && stats_) {
stats_->AddLocalAudioTrack(audio_track().get(), ssrc_);
}
}
void AudioRtpSender::RemoveTrackFromStats() {
if (can_send_track() && stats_) {
stats_->RemoveLocalAudioTrack(audio_track().get(), ssrc_);
}
}
rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
RTC_DCHECK_RUN_ON(signaling_thread_);
return dtmf_sender_proxy_;
}
void AudioRtpSender::SetSend() {
RTC_DCHECK_RUN_ON(signaling_thread_);
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() && audio_track()->GetSource() &&
!audio_track()->GetSource()->remote()) {
options = audio_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 voice_media_channel()->SetAudioSend(ssrc_, track_enabled, &options,
sink_adapter_.get());
});
if (!success) {
RTC_LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc_;
}
}
void AudioRtpSender::ClearSend() {
RTC_DCHECK_RUN_ON(signaling_thread_);
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 voice_media_channel()->SetAudioSend(ssrc_, false, &options, nullptr);
});
if (!success) {
RTC_LOG(LS_WARNING) << "ClearAudioSend: ssrc is incorrect: " << ssrc_;
}
}
rtc::scoped_refptr<VideoRtpSender> VideoRtpSender::Create(
rtc::Thread* worker_thread,
const std::string& id,
SetStreamsObserver* set_streams_observer) {
return rtc::make_ref_counted<VideoRtpSender>(worker_thread, id,
set_streams_observer);
}
VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread,
const std::string& id,
SetStreamsObserver* set_streams_observer)
: RtpSenderBase(worker_thread, id, set_streams_observer) {}
VideoRtpSender::~VideoRtpSender() {
Stop();
}
void VideoRtpSender::OnChanged() {
RTC_DCHECK_RUN_ON(signaling_thread_);
TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
auto content_hint = video_track()->content_hint();
if (cached_track_content_hint_ != content_hint) {
cached_track_content_hint_ = content_hint;
if (can_send_track()) {
SetSend();
}
}
}
void VideoRtpSender::AttachTrack() {
RTC_DCHECK(track_);
cached_track_content_hint_ = video_track()->content_hint();
}
rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
RTC_DCHECK_RUN_ON(signaling_thread_);
RTC_DLOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
return nullptr;
}
void VideoRtpSender::SetSend() {
RTC_DCHECK_RUN_ON(signaling_thread_);
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 = video_track()->GetSource();
if (source) {
options.is_screencast = source->is_screencast();
options.video_noise_reduction = source->needs_denoising();
}
options.content_hint = cached_track_content_hint_;
switch (cached_track_content_hint_) {
case VideoTrackInterface::ContentHint::kNone:
break;
case VideoTrackInterface::ContentHint::kFluid:
options.is_screencast = false;
break;
case VideoTrackInterface::ContentHint::kDetailed:
case VideoTrackInterface::ContentHint::kText:
options.is_screencast = true;
break;
}
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return video_media_channel()->SetVideoSend(ssrc_, &options, video_track());
});
RTC_DCHECK(success);
}
void VideoRtpSender::ClearSend() {
RTC_DCHECK_RUN_ON(signaling_thread_);
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 video_media_channel()->SetVideoSend(ssrc_, nullptr, nullptr);
});
}
} // namespace webrtc