blob: 86026ffd923f4f1fe8830f5e92c7687f2017b450 [file] [log] [blame]
/*
* Copyright (c) 2012 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 "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include <string.h>
#include <memory>
#include <utility>
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_format.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
const char* FrameTypeToString(AudioFrameType frame_type) {
switch (frame_type) {
case AudioFrameType::kEmptyFrame:
return "empty";
case AudioFrameType::kAudioFrameSpeech:
return "audio_speech";
case AudioFrameType::kAudioFrameCN:
return "audio_cn";
}
}
} // namespace
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
: clock_(clock), rtp_sender_(rtp_sender) {
RTC_DCHECK(clock_);
}
RTPSenderAudio::~RTPSenderAudio() {}
int32_t RTPSenderAudio::RegisterAudioPayload(absl::string_view payload_name,
const int8_t payload_type,
const uint32_t frequency,
const size_t channels,
const uint32_t rate) {
if (absl::EqualsIgnoreCase(payload_name, "cn")) {
rtc::CritScope cs(&send_audio_critsect_);
// we can have multiple CNG payload types
switch (frequency) {
case 8000:
cngnb_payload_type_ = payload_type;
break;
case 16000:
cngwb_payload_type_ = payload_type;
break;
case 32000:
cngswb_payload_type_ = payload_type;
break;
case 48000:
cngfb_payload_type_ = payload_type;
break;
default:
return -1;
}
} else if (absl::EqualsIgnoreCase(payload_name, "telephone-event")) {
rtc::CritScope cs(&send_audio_critsect_);
// Don't add it to the list
// we dont want to allow send with a DTMF payloadtype
dtmf_payload_type_ = payload_type;
dtmf_payload_freq_ = frequency;
return 0;
}
return 0;
}
bool RTPSenderAudio::MarkerBit(AudioFrameType frame_type, int8_t payload_type) {
rtc::CritScope cs(&send_audio_critsect_);
// for audio true for first packet in a speech burst
bool marker_bit = false;
if (last_payload_type_ != payload_type) {
if (payload_type != -1 && (cngnb_payload_type_ == payload_type ||
cngwb_payload_type_ == payload_type ||
cngswb_payload_type_ == payload_type ||
cngfb_payload_type_ == payload_type)) {
// Only set a marker bit when we change payload type to a non CNG
return false;
}
// payload_type differ
if (last_payload_type_ == -1) {
if (frame_type != AudioFrameType::kAudioFrameCN) {
// first packet and NOT CNG
return true;
} else {
// first packet and CNG
inband_vad_active_ = true;
return false;
}
}
// not first packet AND
// not CNG AND
// payload_type changed
// set a marker bit when we change payload type
marker_bit = true;
}
// For G.723 G.729, AMR etc we can have inband VAD
if (frame_type == AudioFrameType::kAudioFrameCN) {
inband_vad_active_ = true;
} else if (inband_vad_active_) {
inband_vad_active_ = false;
marker_bit = true;
}
return marker_bit;
}
bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
int8_t payload_type,
uint32_t rtp_timestamp,
const uint8_t* payload_data,
size_t payload_size) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
FrameTypeToString(frame_type));
// From RFC 4733:
// A source has wide latitude as to how often it sends event updates. A
// natural interval is the spacing between non-event audio packets. [...]
// Alternatively, a source MAY decide to use a different spacing for event
// updates, with a value of 50 ms RECOMMENDED.
constexpr int kDtmfIntervalTimeMs = 50;
uint8_t audio_level_dbov = 0;
uint32_t dtmf_payload_freq = 0;
{
rtc::CritScope cs(&send_audio_critsect_);
audio_level_dbov = audio_level_dbov_;
dtmf_payload_freq = dtmf_payload_freq_;
}
// Check if we have pending DTMFs to send
if (!dtmf_event_is_on_ && dtmf_queue_.PendingDtmf()) {
if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) >
kDtmfIntervalTimeMs) {
// New tone to play
dtmf_timestamp_ = rtp_timestamp;
if (dtmf_queue_.NextDtmf(&dtmf_current_event_)) {
dtmf_event_first_packet_sent_ = false;
dtmf_length_samples_ =
dtmf_current_event_.duration_ms * (dtmf_payload_freq / 1000);
dtmf_event_is_on_ = true;
}
}
}
// A source MAY send events and coded audio packets for the same time
// but we don't support it
if (dtmf_event_is_on_) {
if (frame_type == AudioFrameType::kEmptyFrame) {
// kEmptyFrame is used to drive the DTMF when in CN mode
// it can be triggered more frequently than we want to send the
// DTMF packets.
const unsigned int dtmf_interval_time_rtp =
dtmf_payload_freq * kDtmfIntervalTimeMs / 1000;
if ((rtp_timestamp - dtmf_timestamp_last_sent_) <
dtmf_interval_time_rtp) {
// not time to send yet
return true;
}
}
dtmf_timestamp_last_sent_ = rtp_timestamp;
uint32_t dtmf_duration_samples = rtp_timestamp - dtmf_timestamp_;
bool ended = false;
bool send = true;
if (dtmf_length_samples_ > dtmf_duration_samples) {
if (dtmf_duration_samples <= 0) {
// Skip send packet at start, since we shouldn't use duration 0
send = false;
}
} else {
ended = true;
dtmf_event_is_on_ = false;
dtmf_time_last_sent_ = clock_->TimeInMilliseconds();
}
if (send) {
if (dtmf_duration_samples > 0xffff) {
// RFC 4733 2.5.2.3 Long-Duration Events
SendTelephoneEventPacket(ended, dtmf_timestamp_,
static_cast<uint16_t>(0xffff), false);
// set new timestap for this segment
dtmf_timestamp_ = rtp_timestamp;
dtmf_duration_samples -= 0xffff;
dtmf_length_samples_ -= 0xffff;
return SendTelephoneEventPacket(
ended, dtmf_timestamp_,
static_cast<uint16_t>(dtmf_duration_samples), false);
} else {
if (!SendTelephoneEventPacket(ended, dtmf_timestamp_,
dtmf_duration_samples,
!dtmf_event_first_packet_sent_)) {
return false;
}
dtmf_event_first_packet_sent_ = true;
return true;
}
}
return true;
}
if (payload_size == 0 || payload_data == NULL) {
if (frame_type == AudioFrameType::kEmptyFrame) {
// we don't send empty audio RTP packets
// no error since we use it to drive DTMF when we use VAD
return true;
}
return false;
}
std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket();
packet->SetMarker(MarkerBit(frame_type, payload_type));
packet->SetPayloadType(payload_type);
packet->SetTimestamp(rtp_timestamp);
packet->set_capture_time_ms(clock_->TimeInMilliseconds());
// Update audio level extension, if included.
packet->SetExtension<AudioLevel>(
frame_type == AudioFrameType::kAudioFrameSpeech, audio_level_dbov);
uint8_t* payload = packet->AllocatePayload(payload_size);
if (!payload) // Too large payload buffer.
return false;
memcpy(payload, payload_data, payload_size);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;
{
rtc::CritScope cs(&send_audio_critsect_);
last_payload_type_ = payload_type;
}
TRACE_EVENT_ASYNC_END2("webrtc", "Audio", rtp_timestamp, "timestamp",
packet->Timestamp(), "seqnum",
packet->SequenceNumber());
bool send_result = LogAndSendToNetwork(
std::move(packet), kAllowRetransmission, RtpPacketSender::kHighPriority);
if (first_packet_sent_()) {
RTC_LOG(LS_INFO) << "First audio RTP packet sent to pacer";
}
return send_result;
}
// Audio level magnitude and voice activity flag are set for each RTP packet
int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dbov) {
if (level_dbov > 127) {
return -1;
}
rtc::CritScope cs(&send_audio_critsect_);
audio_level_dbov_ = level_dbov;
return 0;
}
// Send a TelephoneEvent tone using RFC 2833 (4733)
int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
uint16_t time_ms,
uint8_t level) {
DtmfQueue::Event event;
{
rtc::CritScope lock(&send_audio_critsect_);
if (dtmf_payload_type_ < 0) {
// TelephoneEvent payloadtype not configured
return -1;
}
event.payload_type = dtmf_payload_type_;
}
event.key = key;
event.duration_ms = time_ms;
event.level = level;
return dtmf_queue_.AddDtmf(event) ? 0 : -1;
}
bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
uint32_t dtmf_timestamp,
uint16_t duration,
bool marker_bit) {
uint8_t send_count = 1;
bool result = true;
if (ended) {
// resend last packet in an event 3 times
send_count = 3;
}
do {
// Send DTMF data.
constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
constexpr size_t kDtmfSize = 4;
std::unique_ptr<RtpPacketToSend> packet(
new RtpPacketToSend(kNoExtensions, kRtpHeaderSize + kDtmfSize));
packet->SetPayloadType(dtmf_current_event_.payload_type);
packet->SetMarker(marker_bit);
packet->SetSsrc(rtp_sender_->SSRC());
packet->SetTimestamp(dtmf_timestamp);
packet->set_capture_time_ms(clock_->TimeInMilliseconds());
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;
// Create DTMF data.
uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize);
RTC_DCHECK(dtmfbuffer);
/* From RFC 2833:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| event |E|R| volume | duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
// R bit always cleared
uint8_t R = 0x00;
uint8_t volume = dtmf_current_event_.level;
// First packet un-ended
uint8_t E = ended ? 0x80 : 0x00;
// First byte is Event number, equals key number
dtmfbuffer[0] = dtmf_current_event_.key;
dtmfbuffer[1] = E | R | volume;
ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
result = LogAndSendToNetwork(std::move(packet), kAllowRetransmission,
RtpPacketSender::kHighPriority);
send_count--;
} while (send_count > 0 && result);
return result;
}
bool RTPSenderAudio::LogAndSendToNetwork(
std::unique_ptr<RtpPacketToSend> packet,
StorageType storage,
RtpPacketSender::Priority priority) {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
int64_t now_ms = clock_->TimeInMilliseconds();
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
rtp_sender_->ActualSendBitrateKbit(),
packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
rtp_sender_->NackOverheadRate() / 1000,
packet->Ssrc());
#endif
return rtp_sender_->SendToNetwork(std::move(packet), storage, priority);
}
} // namespace webrtc