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webrtc/audio/audio_send_stream.cc
Sebastian Jansson cd2a92f8e0 Removes RPLR based FEC controller. 
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.

This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.

Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
2019-10-31 13:56:44 +00:00

883 lines
34 KiB
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/*
* Copyright (c) 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 "audio/audio_send_stream.h"
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/transport.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/function_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/media/media_transport_config.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/conversion.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_audio/vad/include/vad.h"
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/task_queue.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
void UpdateEventLogStreamConfig(RtcEventLog* event_log,
const AudioSendStream::Config& config,
const AudioSendStream::Config* old_config) {
using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
// Only update if any of the things we log have changed.
auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
const absl::optional<SendCodecSpec>& b) {
if (a.has_value() && b.has_value()) {
return a->format.name == b->format.name &&
a->payload_type == b->payload_type;
}
return !a.has_value() && !b.has_value();
};
if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
config.rtp.extensions == old_config->rtp.extensions &&
payload_types_equal(config.send_codec_spec,
old_config->send_codec_spec)) {
return;
}
auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
if (config.send_codec_spec) {
rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
}
event_log->Log(std::make_unique<RtcEventAudioSendStreamConfig>(
std::move(rtclog_config)));
}
} // namespace
constexpr char AudioAllocationConfig::kKey[];
std::unique_ptr<StructParametersParser> AudioAllocationConfig::Parser() {
return StructParametersParser::Create( //
"min", &min_bitrate, //
"max", &max_bitrate, //
"prio_rate", &priority_bitrate, //
"prio_rate_raw", &priority_bitrate_raw, //
"rate_prio", &bitrate_priority);
}
AudioAllocationConfig::AudioAllocationConfig() {
Parser()->Parse(field_trial::FindFullName(kKey));
if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
"exclusive but both were configured.";
}
}
namespace internal {
AudioSendStream::AudioSendStream(
Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state)
: AudioSendStream(clock,
config,
audio_state,
task_queue_factory,
rtp_transport,
bitrate_allocator,
event_log,
rtcp_rtt_stats,
suspended_rtp_state,
voe::CreateChannelSend(clock,
task_queue_factory,
module_process_thread,
config.media_transport_config,
/*overhead_observer=*/this,
config.send_transport,
rtcp_rtt_stats,
event_log,
config.frame_encryptor,
config.crypto_options,
config.rtp.extmap_allow_mixed,
config.rtcp_report_interval_ms,
config.rtp.ssrc)) {}
AudioSendStream::AudioSendStream(
Clock* clock,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
TaskQueueFactory* task_queue_factory,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send)
: clock_(clock),
worker_queue_(rtp_transport->GetWorkerQueue()),
audio_send_side_bwe_(field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")),
allocate_audio_without_feedback_(
field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")),
enable_audio_alr_probing_(
!field_trial::IsDisabled("WebRTC-Audio-AlrProbing")),
send_side_bwe_with_overhead_(
field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
config_(Config(/*send_transport=*/nullptr, MediaTransportConfig())),
audio_state_(audio_state),
channel_send_(std::move(channel_send)),
event_log_(event_log),
use_legacy_overhead_calculation_(
!field_trial::IsDisabled("WebRTC-Audio-LegacyOverhead")),
bitrate_allocator_(bitrate_allocator),
rtp_transport_(rtp_transport),
rtp_rtcp_module_(nullptr),
suspended_rtp_state_(suspended_rtp_state) {
RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
RTC_DCHECK(worker_queue_);
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_send_);
RTC_DCHECK(bitrate_allocator_);
// Currently we require the rtp transport even when media transport is used.
RTC_DCHECK(rtp_transport);
// TODO(nisse): Eventually, we should have only media_transport. But for the
// time being, we can have either. When media transport is injected, there
// should be no rtp_transport, and below check should be strengthened to XOR
// (either rtp_transport or media_transport but not both).
RTC_DCHECK(rtp_transport || config.media_transport_config.media_transport);
if (config.media_transport_config.media_transport) {
// TODO(sukhanov): Currently media transport audio overhead is considered
// constant, we will not get overhead_observer calls when using
// media_transport. In the future when we introduce RTP media transport we
// should make audio overhead interface consistent and work for both RTP and
// non-RTP implementations.
audio_overhead_per_packet_bytes_ =
config.media_transport_config.media_transport->GetAudioPacketOverhead();
}
rtp_rtcp_module_ = channel_send_->GetRtpRtcp();
RTC_DCHECK(rtp_rtcp_module_);
ConfigureStream(config, true);
pacer_thread_checker_.Detach();
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
RTC_DCHECK(!sending_);
channel_send_->ResetSenderCongestionControlObjects();
// Blocking call to synchronize state with worker queue to ensure that there
// are no pending tasks left that keeps references to audio.
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] { thread_sync_event.Set(); });
thread_sync_event.Wait(rtc::Event::kForever);
}
const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
return config_;
}
void AudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& new_config) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
ConfigureStream(new_config, false);
}
AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
const std::vector<RtpExtension>& extensions) {
ExtensionIds ids;
for (const auto& extension : extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
ids.audio_level = extension.id;
} else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
ids.abs_send_time = extension.id;
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
ids.transport_sequence_number = extension.id;
} else if (extension.uri == RtpExtension::kMidUri) {
ids.mid = extension.id;
} else if (extension.uri == RtpExtension::kRidUri) {
ids.rid = extension.id;
} else if (extension.uri == RtpExtension::kRepairedRidUri) {
ids.repaired_rid = extension.id;
}
}
return ids;
}
int AudioSendStream::TransportSeqNumId(const AudioSendStream::Config& config) {
return FindExtensionIds(config.rtp.extensions).transport_sequence_number;
}
void AudioSendStream::ConfigureStream(
const webrtc::AudioSendStream::Config& new_config,
bool first_time) {
RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
<< new_config.ToString();
UpdateEventLogStreamConfig(event_log_, new_config,
first_time ? nullptr : &config_);
const auto& old_config = config_;
// Configuration parameters which cannot be changed.
RTC_DCHECK(first_time ||
old_config.send_transport == new_config.send_transport);
RTC_DCHECK(first_time || old_config.rtp.ssrc == new_config.rtp.ssrc);
if (suspended_rtp_state_ && first_time) {
rtp_rtcp_module_->SetRtpState(*suspended_rtp_state_);
}
if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
channel_send_->SetRTCP_CNAME(new_config.rtp.c_name);
}
// Enable the frame encryptor if a new frame encryptor has been provided.
if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
channel_send_->SetFrameEncryptor(new_config.frame_encryptor);
}
if (first_time ||
new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) {
channel_send_->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed);
}
const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
// Audio level indication
if (first_time || new_ids.audio_level != old_ids.audio_level) {
channel_send_->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
new_ids.audio_level);
}
if (first_time || new_ids.abs_send_time != old_ids.abs_send_time) {
channel_send_->GetRtpRtcp()->DeregisterSendRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime);
if (new_ids.abs_send_time) {
rtp_rtcp_module_->RegisterRtpHeaderExtension(AbsoluteSendTime::kUri,
new_ids.abs_send_time);
}
}
bool transport_seq_num_id_changed =
new_ids.transport_sequence_number != old_ids.transport_sequence_number;
if (first_time ||
(transport_seq_num_id_changed && !allocate_audio_without_feedback_)) {
if (!first_time) {
channel_send_->ResetSenderCongestionControlObjects();
}
RtcpBandwidthObserver* bandwidth_observer = nullptr;
if (audio_send_side_bwe_ && !allocate_audio_without_feedback_ &&
new_ids.transport_sequence_number != 0) {
channel_send_->EnableSendTransportSequenceNumber(
new_ids.transport_sequence_number);
// Probing in application limited region is only used in combination with
// send side congestion control, wich depends on feedback packets which
// requires transport sequence numbers to be enabled.
// Optionally request ALR probing but do not override any existing
// request from other streams.
if (enable_audio_alr_probing_) {
rtp_transport_->EnablePeriodicAlrProbing(true);
}
bandwidth_observer = rtp_transport_->GetBandwidthObserver();
}
channel_send_->RegisterSenderCongestionControlObjects(rtp_transport_,
bandwidth_observer);
}
// MID RTP header extension.
if ((first_time || new_ids.mid != old_ids.mid ||
new_config.rtp.mid != old_config.rtp.mid) &&
new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
channel_send_->SetMid(new_config.rtp.mid, new_ids.mid);
}
// RID RTP header extension
if ((first_time || new_ids.rid != old_ids.rid ||
new_ids.repaired_rid != old_ids.repaired_rid ||
new_config.rtp.rid != old_config.rtp.rid)) {
channel_send_->SetRid(new_config.rtp.rid, new_ids.rid,
new_ids.repaired_rid);
}
if (!ReconfigureSendCodec(new_config)) {
RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
}
if (sending_) {
ReconfigureBitrateObserver(new_config);
}
config_ = new_config;
}
void AudioSendStream::Start() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (sending_) {
return;
}
if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
config_.max_bitrate_bps != -1 &&
(allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(worker_queue_);
ConfigureBitrateObserver();
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
} else {
rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
channel_send_->StartSend();
sending_ = true;
audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
encoder_num_channels_);
}
void AudioSendStream::Stop() {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
if (!sending_) {
return;
}
RemoveBitrateObserver();
channel_send_->StopSend();
sending_ = false;
audio_state()->RemoveSendingStream(this);
}
void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK_GT(audio_frame->sample_rate_hz_, 0);
double duration = static_cast<double>(audio_frame->samples_per_channel_) /
audio_frame->sample_rate_hz_;
{
// Note: SendAudioData() passes the frame further down the pipeline and it
// may eventually get sent. But this method is invoked even if we are not
// connected, as long as we have an AudioSendStream (created as a result of
// an O/A exchange). This means that we are calculating audio levels whether
// or not we are sending samples.
// TODO(https://crbug.com/webrtc/10771): All "media-source" related stats
// should move from send-streams to the local audio sources or tracks; a
// send-stream should not be required to read the microphone audio levels.
rtc::CritScope cs(&audio_level_lock_);
audio_level_.ComputeLevel(*audio_frame, duration);
}
channel_send_->ProcessAndEncodeAudio(std::move(audio_frame));
}
bool AudioSendStream::SendTelephoneEvent(int payload_type,
int payload_frequency,
int event,
int duration_ms) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
channel_send_->SetSendTelephoneEventPayloadType(payload_type,
payload_frequency);
return channel_send_->SendTelephoneEventOutband(event, duration_ms);
}
void AudioSendStream::SetMuted(bool muted) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
channel_send_->SetInputMute(muted);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
return GetStats(true);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
bool has_remote_tracks) const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = config_.rtp.ssrc;
stats.target_bitrate_bps = channel_send_->GetBitrate();
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
stats.payload_bytes_sent = call_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
call_stats.header_and_padding_bytes_sent;
stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
stats.packets_sent = call_stats.packetsSent;
stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
if (call_stats.rttMs > 0) {
stats.rtt_ms = call_stats.rttMs;
}
if (config_.send_codec_spec) {
const auto& spec = *config_.send_codec_spec;
stats.codec_name = spec.format.name;
stats.codec_payload_type = spec.payload_type;
// Get data from the last remote RTCP report.
for (const auto& block : channel_send_->GetRemoteRTCPReportBlocks()) {
// Lookup report for send ssrc only.
if (block.source_SSRC == stats.local_ssrc) {
stats.packets_lost = block.cumulative_num_packets_lost;
stats.fraction_lost = Q8ToFloat(block.fraction_lost);
// Convert timestamps to milliseconds.
if (spec.format.clockrate_hz / 1000 > 0) {
stats.jitter_ms =
block.interarrival_jitter / (spec.format.clockrate_hz / 1000);
}
break;
}
}
}
{
rtc::CritScope cs(&audio_level_lock_);
stats.audio_level = audio_level_.LevelFullRange();
stats.total_input_energy = audio_level_.TotalEnergy();
stats.total_input_duration = audio_level_.TotalDuration();
}
stats.typing_noise_detected = audio_state()->typing_noise_detected();
stats.ana_statistics = channel_send_->GetANAStatistics();
RTC_DCHECK(audio_state_->audio_processing());
stats.apm_statistics =
audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
stats.report_block_datas = std::move(call_stats.report_block_datas);
return stats;
}
void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!worker_thread_checker_.IsCurrent());
channel_send_->ReceivedRTCPPacket(packet, length);
}
uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
RTC_DCHECK_RUN_ON(worker_queue_);
// Pick a target bitrate between the constraints. Overrules the allocator if
// it 1) allocated a bitrate of zero to disable the stream or 2) allocated a
// higher than max to allow for e.g. extra FEC.
auto constraints = GetMinMaxBitrateConstraints();
update.target_bitrate.Clamp(constraints.min, constraints.max);
channel_send_->OnBitrateAllocation(update);
// The amount of audio protection is not exposed by the encoder, hence
// always returning 0.
return 0;
}
void AudioSendStream::SetTransportOverhead(
int transport_overhead_per_packet_bytes) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
rtc::CritScope cs(&overhead_per_packet_lock_);
transport_overhead_per_packet_bytes_ = transport_overhead_per_packet_bytes;
UpdateOverheadForEncoder();
}
void AudioSendStream::OnOverheadChanged(
size_t overhead_bytes_per_packet_bytes) {
rtc::CritScope cs(&overhead_per_packet_lock_);
audio_overhead_per_packet_bytes_ = overhead_bytes_per_packet_bytes;
UpdateOverheadForEncoder();
}
void AudioSendStream::UpdateOverheadForEncoder() {
const size_t overhead_per_packet_bytes = GetPerPacketOverheadBytes();
if (overhead_per_packet_bytes == 0) {
return; // Overhead is not known yet, do not tell the encoder.
}
channel_send_->CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedOverhead(overhead_per_packet_bytes);
});
worker_queue_->PostTask([this, overhead_per_packet_bytes] {
RTC_DCHECK_RUN_ON(worker_queue_);
if (total_packet_overhead_bytes_ != overhead_per_packet_bytes) {
total_packet_overhead_bytes_ = overhead_per_packet_bytes;
if (registered_with_allocator_) {
ConfigureBitrateObserver();
}
}
});
}
size_t AudioSendStream::TestOnlyGetPerPacketOverheadBytes() const {
rtc::CritScope cs(&overhead_per_packet_lock_);
return GetPerPacketOverheadBytes();
}
size_t AudioSendStream::GetPerPacketOverheadBytes() const {
return transport_overhead_per_packet_bytes_ +
audio_overhead_per_packet_bytes_;
}
RtpState AudioSendStream::GetRtpState() const {
return rtp_rtcp_module_->GetRtpState();
}
const voe::ChannelSendInterface* AudioSendStream::GetChannel() const {
return channel_send_.get();
}
internal::AudioState* AudioSendStream::audio_state() {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
const internal::AudioState* AudioSendStream::audio_state() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
void AudioSendStream::StoreEncoderProperties(int sample_rate_hz,
size_t num_channels) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
encoder_sample_rate_hz_ = sample_rate_hz;
encoder_num_channels_ = num_channels;
if (sending_) {
// Update AudioState's information about the stream.
audio_state()->AddSendingStream(this, sample_rate_hz, num_channels);
}
}
// Apply current codec settings to a single voe::Channel used for sending.
bool AudioSendStream::SetupSendCodec(const Config& new_config) {
RTC_DCHECK(new_config.send_codec_spec);
const auto& spec = *new_config.send_codec_spec;
RTC_DCHECK(new_config.encoder_factory);
std::unique_ptr<AudioEncoder> encoder =
new_config.encoder_factory->MakeAudioEncoder(
spec.payload_type, spec.format, new_config.codec_pair_id);
if (!encoder) {
RTC_DLOG(LS_ERROR) << "Unable to create encoder for "
<< rtc::ToString(spec.format);
return false;
}
// If a bitrate has been specified for the codec, use it over the
// codec's default.
if (spec.target_bitrate_bps) {
encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
}
// Enable ANA if configured (currently only used by Opus).
if (new_config.audio_network_adaptor_config) {
if (encoder->EnableAudioNetworkAdaptor(
*new_config.audio_network_adaptor_config, event_log_)) {
RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
<< new_config.rtp.ssrc;
} else {
RTC_NOTREACHED();
}
}
// Wrap the encoder in a an AudioEncoderCNG, if VAD is enabled.
if (spec.cng_payload_type) {
AudioEncoderCngConfig cng_config;
cng_config.num_channels = encoder->NumChannels();
cng_config.payload_type = *spec.cng_payload_type;
cng_config.speech_encoder = std::move(encoder);
cng_config.vad_mode = Vad::kVadNormal;
encoder = CreateComfortNoiseEncoder(std::move(cng_config));
RegisterCngPayloadType(*spec.cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
// Set currently known overhead (used in ANA, opus only).
// If overhead changes later, it will be updated in UpdateOverheadForEncoder.
{
rtc::CritScope cs(&overhead_per_packet_lock_);
if (GetPerPacketOverheadBytes() > 0) {
encoder->OnReceivedOverhead(GetPerPacketOverheadBytes());
}
}
worker_queue_->PostTask(
[this, length_range = encoder->GetFrameLengthRange()] {
RTC_DCHECK_RUN_ON(worker_queue_);
frame_length_range_ = length_range;
});
StoreEncoderProperties(encoder->SampleRateHz(), encoder->NumChannels());
channel_send_->SetEncoder(new_config.send_codec_spec->payload_type,
std::move(encoder));
return true;
}
bool AudioSendStream::ReconfigureSendCodec(const Config& new_config) {
const auto& old_config = config_;
if (!new_config.send_codec_spec) {
// We cannot de-configure a send codec. So we will do nothing.
// By design, the send codec should have not been configured.
RTC_DCHECK(!old_config.send_codec_spec);
return true;
}
if (new_config.send_codec_spec == old_config.send_codec_spec &&
new_config.audio_network_adaptor_config ==
old_config.audio_network_adaptor_config) {
return true;
}
// If we have no encoder, or the format or payload type's changed, create a
// new encoder.
if (!old_config.send_codec_spec ||
new_config.send_codec_spec->format !=
old_config.send_codec_spec->format ||
new_config.send_codec_spec->payload_type !=
old_config.send_codec_spec->payload_type) {
return SetupSendCodec(new_config);
}
const absl::optional<int>& new_target_bitrate_bps =
new_config.send_codec_spec->target_bitrate_bps;
// If a bitrate has been specified for the codec, use it over the
// codec's default.
if (new_target_bitrate_bps &&
new_target_bitrate_bps !=
old_config.send_codec_spec->target_bitrate_bps) {
channel_send_->CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
});
}
ReconfigureANA(new_config);
ReconfigureCNG(new_config);
// Set currently known overhead (used in ANA, opus only).
{
rtc::CritScope cs(&overhead_per_packet_lock_);
UpdateOverheadForEncoder();
}
return true;
}
void AudioSendStream::ReconfigureANA(const Config& new_config) {
if (new_config.audio_network_adaptor_config ==
config_.audio_network_adaptor_config) {
return;
}
if (new_config.audio_network_adaptor_config) {
channel_send_->CallEncoder([&](AudioEncoder* encoder) {
if (encoder->EnableAudioNetworkAdaptor(
*new_config.audio_network_adaptor_config, event_log_)) {
RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
<< new_config.rtp.ssrc;
} else {
RTC_NOTREACHED();
}
});
} else {
channel_send_->CallEncoder(
[&](AudioEncoder* encoder) { encoder->DisableAudioNetworkAdaptor(); });
RTC_DLOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
<< new_config.rtp.ssrc;
}
}
void AudioSendStream::ReconfigureCNG(const Config& new_config) {
if (new_config.send_codec_spec->cng_payload_type ==
config_.send_codec_spec->cng_payload_type) {
return;
}
// Register the CNG payload type if it's been added, don't do anything if CNG
// is removed. Payload types must not be redefined.
if (new_config.send_codec_spec->cng_payload_type) {
RegisterCngPayloadType(*new_config.send_codec_spec->cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
// Wrap or unwrap the encoder in an AudioEncoderCNG.
channel_send_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
auto sub_encoders = old_encoder->ReclaimContainedEncoders();
if (!sub_encoders.empty()) {
// Replace enc with its sub encoder. We need to put the sub
// encoder in a temporary first, since otherwise the old value
// of enc would be destroyed before the new value got assigned,
// which would be bad since the new value is a part of the old
// value.
auto tmp = std::move(sub_encoders[0]);
old_encoder = std::move(tmp);
}
if (new_config.send_codec_spec->cng_payload_type) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(old_encoder);
config.num_channels = config.speech_encoder->NumChannels();
config.payload_type = *new_config.send_codec_spec->cng_payload_type;
config.vad_mode = Vad::kVadNormal;
*encoder_ptr = CreateComfortNoiseEncoder(std::move(config));
} else {
*encoder_ptr = std::move(old_encoder);
}
});
}
void AudioSendStream::ReconfigureBitrateObserver(
const webrtc::AudioSendStream::Config& new_config) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Since the Config's default is for both of these to be -1, this test will
// allow us to configure the bitrate observer if the new config has bitrate
// limits set, but would only have us call RemoveBitrateObserver if we were
// previously configured with bitrate limits.
if (config_.min_bitrate_bps == new_config.min_bitrate_bps &&
config_.max_bitrate_bps == new_config.max_bitrate_bps &&
config_.bitrate_priority == new_config.bitrate_priority &&
(TransportSeqNumId(config_) == TransportSeqNumId(new_config) ||
!audio_send_side_bwe_)) {
return;
}
if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(worker_queue_);
registered_with_allocator_ = true;
// We may get a callback immediately as the observer is registered, so
// make
// sure the bitrate limits in config_ are up-to-date.
config_.min_bitrate_bps = new_config.min_bitrate_bps;
config_.max_bitrate_bps = new_config.max_bitrate_bps;
config_.bitrate_priority = new_config.bitrate_priority;
ConfigureBitrateObserver();
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
rtp_transport_->AccountForAudioPacketsInPacedSender(false);
RemoveBitrateObserver();
rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
}
void AudioSendStream::ConfigureBitrateObserver() {
// This either updates the current observer or adds a new observer.
// TODO(srte): Add overhead compensation here.
auto constraints = GetMinMaxBitrateConstraints();
DataRate priority_bitrate = allocation_settings_.priority_bitrate;
if (send_side_bwe_with_overhead_) {
if (use_legacy_overhead_calculation_) {
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
const TimeDelta kMinPacketDuration = TimeDelta::ms(20);
DataRate max_overhead =
DataSize::bytes(kOverheadPerPacket) / kMinPacketDuration;
priority_bitrate += max_overhead;
} else {
RTC_DCHECK(frame_length_range_);
const DataSize kOverheadPerPacket =
DataSize::bytes(total_packet_overhead_bytes_);
DataRate max_overhead = kOverheadPerPacket / frame_length_range_->first;
priority_bitrate += max_overhead;
}
}
if (allocation_settings_.priority_bitrate_raw)
priority_bitrate = *allocation_settings_.priority_bitrate_raw;
bitrate_allocator_->AddObserver(
this,
MediaStreamAllocationConfig{
constraints.min.bps<uint32_t>(), constraints.max.bps<uint32_t>(), 0,
priority_bitrate.bps(), true,
allocation_settings_.bitrate_priority.value_or(
config_.bitrate_priority)});
}
void AudioSendStream::RemoveBitrateObserver() {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
rtc::Event thread_sync_event;
worker_queue_->PostTask([this, &thread_sync_event] {
RTC_DCHECK_RUN_ON(worker_queue_);
registered_with_allocator_ = false;
bitrate_allocator_->RemoveObserver(this);
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
}
AudioSendStream::TargetAudioBitrateConstraints
AudioSendStream::GetMinMaxBitrateConstraints() const {
TargetAudioBitrateConstraints constraints{
DataRate::bps(config_.min_bitrate_bps),
DataRate::bps(config_.max_bitrate_bps)};
// If bitrates were explicitly overriden via field trial, use those values.
if (allocation_settings_.min_bitrate)
constraints.min = *allocation_settings_.min_bitrate;
if (allocation_settings_.max_bitrate)
constraints.max = *allocation_settings_.max_bitrate;
RTC_DCHECK_GE(constraints.min, DataRate::Zero());
RTC_DCHECK_GE(constraints.max, DataRate::Zero());
RTC_DCHECK_GE(constraints.max, constraints.min);
if (send_side_bwe_with_overhead_) {
if (use_legacy_overhead_calculation_) {
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
const DataSize kOverheadPerPacket = DataSize::bytes(20 + 8 + 10 + 12);
const TimeDelta kMaxFrameLength =
TimeDelta::ms(60); // Based on Opus spec
const DataRate kMinOverhead = kOverheadPerPacket / kMaxFrameLength;
constraints.min += kMinOverhead;
constraints.max += kMinOverhead;
} else {
RTC_DCHECK(frame_length_range_);
const DataSize kOverheadPerPacket =
DataSize::bytes(total_packet_overhead_bytes_);
constraints.min += kOverheadPerPacket / frame_length_range_->second;
constraints.max += kOverheadPerPacket / frame_length_range_->first;
}
}
return constraints;
}
void AudioSendStream::RegisterCngPayloadType(int payload_type,
int clockrate_hz) {
channel_send_->RegisterCngPayloadType(payload_type, clockrate_hz);
}
} // namespace internal
} // namespace webrtc
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