blob: 417720cdb0a0bb264bd07d04aab5aa5cb451b8e9 [file] [log] [blame]
/*
* 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 "webrtc/audio/audio_send_stream.h"
#include <string>
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrc: " << ssrc;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << ", nack: " << nack.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", voe_channel_id: " << voe_channel_id;
// TODO(solenberg): Encoder config.
ss << ", cng_payload_type: " << cng_payload_type;
ss << '}';
return ss.str();
}
namespace internal {
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
CongestionController* congestion_controller,
BitrateAllocator* bitrate_allocator)
: worker_queue_(worker_queue),
config_(config),
audio_state_(audio_state),
bitrate_allocator_(bitrate_allocator) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(congestion_controller);
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->RegisterSenderCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(),
congestion_controller->packet_router());
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
config_.rtp.nack.rtp_history_ms / 20);
channel_proxy_->RegisterExternalTransport(config.send_transport);
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAbsSendTimeUri) {
channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kAudioLevelUri) {
channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";
}
}
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
}
void AudioSendStream::Start() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
rtc::Event thread_sync_event(false /* manual_reset */, false);
worker_queue_->PostTask([this, &thread_sync_event] {
bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
config_.max_bitrate_kbps * 1000, 0, true);
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
}
ScopedVoEInterface<VoEBase> base(voice_engine());
int error = base->StartSend(config_.voe_channel_id);
if (error != 0) {
LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
}
}
void AudioSendStream::Stop() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
rtc::Event thread_sync_event(false /* manual_reset */, false);
worker_queue_->PostTask([this, &thread_sync_event] {
bitrate_allocator_->RemoveObserver(this);
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
ScopedVoEInterface<VoEBase> base(voice_engine());
int error = base->StopSend(config_.voe_channel_id);
if (error != 0) {
LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error;
}
}
bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
int duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
}
void AudioSendStream::SetMuted(bool muted) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel_proxy_->SetInputMute(muted);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = config_.rtp.ssrc;
ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
ScopedVoEInterface<VoECodec> codec(voice_engine());
ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
stats.bytes_sent = call_stats.bytesSent;
stats.packets_sent = call_stats.packetsSent;
// 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;
}
// TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
// implementation.
stats.aec_quality_min = -1;
webrtc::CodecInst codec_inst = {0};
if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
RTC_DCHECK_NE(codec_inst.pltype, -1);
stats.codec_name = codec_inst.plname;
// Get data from the last remote RTCP report.
for (const auto& block : channel_proxy_->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);
stats.ext_seqnum = block.extended_highest_sequence_number;
// Convert samples to milliseconds.
if (codec_inst.plfreq / 1000 > 0) {
stats.jitter_ms =
block.interarrival_jitter / (codec_inst.plfreq / 1000);
}
break;
}
}
}
// Local speech level.
{
unsigned int level = 0;
int error = volume->GetSpeechInputLevelFullRange(level);
RTC_DCHECK_EQ(0, error);
stats.audio_level = static_cast<int32_t>(level);
}
bool echo_metrics_on = false;
int error = processing->GetEcMetricsStatus(echo_metrics_on);
RTC_DCHECK_EQ(0, error);
if (echo_metrics_on) {
// These can also be negative, but in practice -1 is only used to signal
// insufficient data, since the resolution is limited to multiples of 4 ms.
int median = -1;
int std = -1;
float dummy = 0.0f;
error = processing->GetEcDelayMetrics(median, std, dummy);
RTC_DCHECK_EQ(0, error);
stats.echo_delay_median_ms = median;
stats.echo_delay_std_ms = std;
// These can take on valid negative values, so use the lowest possible level
// as default rather than -1.
int erl = -100;
int erle = -100;
int dummy1 = 0;
int dummy2 = 0;
error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
RTC_DCHECK_EQ(0, error);
stats.echo_return_loss = erl;
stats.echo_return_loss_enhancement = erle;
}
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
stats.typing_noise_detected = audio_state->typing_noise_detected();
return stats;
}
void AudioSendStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
bool 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(!thread_checker_.CalledOnValidThread());
return channel_proxy_->ReceivedRTCPPacket(packet, length);
}
uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt) {
RTC_DCHECK_GE(bitrate_bps,
static_cast<uint32_t>(config_.min_bitrate_kbps * 1000));
// The bitrate allocator might allocate an higher than max configured bitrate
// if there is room, to allow for, as example, extra FEC. Ignore that for now.
const uint32_t max_bitrate_bps = config_.max_bitrate_kbps * 1000;
if (bitrate_bps > max_bitrate_bps)
bitrate_bps = max_bitrate_bps;
channel_proxy_->SetBitrate(bitrate_bps);
// The amount of audio protection is not exposed by the encoder, hence
// always returning 0.
return 0;
}
const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;
}
VoiceEngine* AudioSendStream::voice_engine() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
VoiceEngine* voice_engine = audio_state->voice_engine();
RTC_DCHECK(voice_engine);
return voice_engine;
}
} // namespace internal
} // namespace webrtc