blob: 560055d4f4fa50bb9829bd292753b7df1f033b94 [file] [log] [blame]
/*
* Copyright (c) 2020 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/voip/audio_ingress.h"
#include <algorithm>
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_format.h"
#include "audio/utility/audio_frame_operations.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace {
AudioCodingModule::Config CreateAcmConfig(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
AudioCodingModule::Config acm_config;
acm_config.neteq_config.enable_muted_state = true;
acm_config.decoder_factory = decoder_factory;
return acm_config;
}
} // namespace
AudioIngress::AudioIngress(
RtpRtcpInterface* rtp_rtcp,
Clock* clock,
ReceiveStatistics* receive_statistics,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: playing_(false),
remote_ssrc_(0),
first_rtp_timestamp_(-1),
rtp_receive_statistics_(receive_statistics),
rtp_rtcp_(rtp_rtcp),
acm_receiver_(CreateAcmConfig(decoder_factory)),
ntp_estimator_(clock) {}
AudioIngress::~AudioIngress() = default;
AudioMixer::Source::AudioFrameInfo AudioIngress::GetAudioFrameWithInfo(
int sampling_rate,
AudioFrame* audio_frame) {
audio_frame->sample_rate_hz_ = sampling_rate;
// Get 10ms raw PCM data from the ACM.
bool muted = false;
if (acm_receiver_.GetAudio(sampling_rate, audio_frame, &muted) == -1) {
RTC_DLOG(LS_ERROR) << "GetAudio() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
// irrelevant.
return AudioMixer::Source::AudioFrameInfo::kError;
}
if (muted) {
AudioFrameOperations::Mute(audio_frame);
}
// Measure audio level.
constexpr double kAudioSampleDurationSeconds = 0.01;
output_audio_level_.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
// Set first rtp timestamp with first audio frame with valid timestamp.
if (first_rtp_timestamp_ < 0 && audio_frame->timestamp_ != 0) {
first_rtp_timestamp_ = audio_frame->timestamp_;
}
if (first_rtp_timestamp_ >= 0) {
// Compute elapsed and NTP times.
int64_t unwrap_timestamp;
{
MutexLock lock(&lock_);
unwrap_timestamp =
timestamp_wrap_handler_.Unwrap(audio_frame->timestamp_);
audio_frame->ntp_time_ms_ =
ntp_estimator_.Estimate(audio_frame->timestamp_);
}
// For clock rate, default to the playout sampling rate if we haven't
// received any packets yet.
absl::optional<std::pair<int, SdpAudioFormat>> decoder =
acm_receiver_.LastDecoder();
int clock_rate = decoder ? decoder->second.clockrate_hz
: acm_receiver_.last_output_sample_rate_hz();
RTC_DCHECK_GT(clock_rate, 0);
audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - first_rtp_timestamp_) / (clock_rate / 1000);
}
return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
: AudioMixer::Source::AudioFrameInfo::kNormal;
}
void AudioIngress::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
{
MutexLock lock(&lock_);
for (const auto& kv : codecs) {
receive_codec_info_[kv.first] = kv.second.clockrate_hz;
}
}
acm_receiver_.SetCodecs(codecs);
}
void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
if (!IsPlaying()) {
return;
}
RtpPacketReceived rtp_packet_received;
rtp_packet_received.Parse(rtp_packet.data(), rtp_packet.size());
// Set payload type's sampling rate before we feed it into ReceiveStatistics.
{
MutexLock lock(&lock_);
const auto& it =
receive_codec_info_.find(rtp_packet_received.PayloadType());
// If sampling rate info is not available in our received codec set, it
// would mean that remote media endpoint is sending incorrect payload id
// which can't be processed correctly especially on payload type id in
// dynamic range.
if (it == receive_codec_info_.end()) {
RTC_DLOG(LS_WARNING) << "Unexpected payload id received: "
<< rtp_packet_received.PayloadType();
return;
}
rtp_packet_received.set_payload_type_frequency(it->second);
}
rtp_receive_statistics_->OnRtpPacket(rtp_packet_received);
RTPHeader header;
rtp_packet_received.GetHeader(&header);
size_t packet_length = rtp_packet_received.size();
if (packet_length < header.headerLength ||
(packet_length - header.headerLength) < header.paddingLength) {
RTC_DLOG(LS_ERROR) << "Packet length(" << packet_length << ") header("
<< header.headerLength << ") padding("
<< header.paddingLength << ")";
return;
}
const uint8_t* payload = rtp_packet_received.data() + header.headerLength;
size_t payload_length = packet_length - header.headerLength;
size_t payload_data_length = payload_length - header.paddingLength;
auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
// Push the incoming payload (parsed and ready for decoding) into the ACM.
if (acm_receiver_.InsertPacket(header, data_view) != 0) {
RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
"push data to the ACM";
}
}
void AudioIngress::ReceivedRTCPPacket(
rtc::ArrayView<const uint8_t> rtcp_packet) {
// Deliver RTCP packet to RTP/RTCP module for parsing.
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet.data(), rtcp_packet.size());
int64_t rtt = GetRoundTripTime();
if (rtt == -1) {
// Waiting for valid RTT.
return;
}
uint32_t ntp_secs = 0, ntp_frac = 0, rtp_timestamp = 0;
if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
&rtp_timestamp) != 0) {
// Waiting for RTCP.
return;
}
{
MutexLock lock(&lock_);
ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
}
}
int64_t AudioIngress::GetRoundTripTime() {
const std::vector<ReportBlockData>& report_data =
rtp_rtcp_->GetLatestReportBlockData();
// If we do not have report block which means remote RTCP hasn't be received
// yet, return -1 as to indicate uninitialized value.
if (report_data.empty()) {
return -1;
}
// We don't know in advance the remote SSRC used by the other end's receiver
// reports, so use the SSRC of the first report block as remote SSRC for now.
// TODO(natim@webrtc.org): handle the case where remote end is changing ssrc
// and update accordingly here.
const ReportBlockData& block_data = report_data[0];
const uint32_t sender_ssrc = block_data.report_block().sender_ssrc;
if (sender_ssrc != remote_ssrc_.load()) {
remote_ssrc_.store(sender_ssrc);
rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
}
return (block_data.has_rtt() ? block_data.last_rtt_ms() : -1);
}
} // namespace webrtc