admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
magjed 56124bd158 Send audio and video codecs to RTPPayloadRegistry 
The purpose with this CL is to be able to send video codec specific
information down to RTPPayloadRegistry. We already do this for audio
with explicit arguments for e.g. number of channels. Instead of
extracting the arguments from webrtc::CodecInst (audio) and
webrtc::VideoCodec, this CL sends the types unmodified all the way down
to RTPPayloadRegistry.

This CL does not contain any functional changes, and is just a
preparation for future CL:s.

In the dependent CL https://codereview.webrtc.org/2524923002/,
RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled
audio/video specific aspects of payload handling. After this CL, we will
know if we get audio or video codecs without any dependency injection,
since we have different functions with different signatures for audio
vs video.

BUG=webrtc:6743
TBR=mflodman

Review-Url: https://codereview.webrtc.org/2523843002
Cr-Commit-Position: refs/heads/master@{#15231}
2016-11-24 17:34:53 +00:00

308 lines
11 KiB
C++
Raw Blame History

/*
* 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 "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
#include <assert.h> // assert
#include <math.h> // pow()
#include <string.h> // memcpy()
#include "webrtc/common_types.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
namespace webrtc {
RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
RtpData* data_callback) {
return new RTPReceiverAudio(data_callback);
}
RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
: RTPReceiverStrategy(data_callback),
TelephoneEventHandler(),
telephone_event_forward_to_decoder_(false),
telephone_event_payload_type_(-1),
cng_nb_payload_type_(-1),
cng_wb_payload_type_(-1),
cng_swb_payload_type_(-1),
cng_fb_payload_type_(-1),
num_energy_(0),
current_remote_energy_() {
last_payload_.Audio.channels = 1;
memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
}
// Outband TelephoneEvent(DTMF) detection
void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
bool forward_to_decoder) {
rtc::CritScope lock(&crit_sect_);
telephone_event_forward_to_decoder_ = forward_to_decoder;
}
// Is forwarding of outband telephone events turned on/off?
bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
rtc::CritScope lock(&crit_sect_);
return telephone_event_forward_to_decoder_;
}
bool RTPReceiverAudio::TelephoneEventPayloadType(
int8_t payload_type) const {
rtc::CritScope lock(&crit_sect_);
return telephone_event_payload_type_ == payload_type;
}
bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) {
rtc::CritScope lock(&crit_sect_);
return payload_type == cng_nb_payload_type_ ||
payload_type == cng_wb_payload_type_ ||
payload_type == cng_swb_payload_type_ ||
payload_type == cng_fb_payload_type_;
}
bool RTPReceiverAudio::ShouldReportCsrcChanges(uint8_t payload_type) const {
// Don't do this for DTMF packets, otherwise it's fine.
return !TelephoneEventPayloadType(payload_type);
}
// - Sample based or frame based codecs based on RFC 3551
// -
// - NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
// - The correct rate is 4 bits/sample.
// -
// - name of sampling default
// - encoding sample/frame bits/sample rate ms/frame ms/packet
// -
// - Sample based audio codecs
// - DVI4 sample 4 var. 20
// - G722 sample 4 16,000 20
// - G726-40 sample 5 8,000 20
// - G726-32 sample 4 8,000 20
// - G726-24 sample 3 8,000 20
// - G726-16 sample 2 8,000 20
// - L8 sample 8 var. 20
// - L16 sample 16 var. 20
// - PCMA sample 8 var. 20
// - PCMU sample 8 var. 20
// -
// - Frame based audio codecs
// - G723 frame N/A 8,000 30 30
// - G728 frame N/A 8,000 2.5 20
// - G729 frame N/A 8,000 10 20
// - G729D frame N/A 8,000 10 20
// - G729E frame N/A 8,000 10 20
// - GSM frame N/A 8,000 20 20
// - GSM-EFR frame N/A 8,000 20 20
// - LPC frame N/A 8,000 20 20
// - MPA frame N/A var. var.
// -
// - G7221 frame N/A
int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
const CodecInst& audio_codec) {
rtc::CritScope lock(&crit_sect_);
if (RtpUtility::StringCompare(audio_codec.plname, "telephone-event", 15)) {
telephone_event_payload_type_ = audio_codec.pltype;
}
if (RtpUtility::StringCompare(audio_codec.plname, "cn", 2)) {
// We support comfort noise at four different frequencies.
if (audio_codec.plfreq == 8000) {
cng_nb_payload_type_ = audio_codec.pltype;
} else if (audio_codec.plfreq == 16000) {
cng_wb_payload_type_ = audio_codec.pltype;
} else if (audio_codec.plfreq == 32000) {
cng_swb_payload_type_ = audio_codec.pltype;
} else if (audio_codec.plfreq == 48000) {
cng_fb_payload_type_ = audio_codec.pltype;
} else {
assert(false);
return -1;
}
}
return 0;
}
int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* payload,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) {
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::ParseRtp",
"seqnum", rtp_header->header.sequenceNumber, "timestamp",
rtp_header->header.timestamp);
rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs;
num_energy_ = rtp_header->type.Audio.numEnergy;
if (rtp_header->type.Audio.numEnergy > 0 &&
rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
memcpy(current_remote_energy_,
rtp_header->type.Audio.arrOfEnergy,
rtp_header->type.Audio.numEnergy);
}
if (first_packet_received_()) {
LOG(LS_INFO) << "Received first audio RTP packet";
}
return ParseAudioCodecSpecific(rtp_header,
payload,
payload_length,
specific_payload.Audio,
is_red);
}
RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
uint16_t last_payload_length) const {
// Our CNG is 9 bytes; if it's a likely CNG the receiver needs to check
// kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG.
if (last_payload_length < 10) { // our CNG is 9 bytes
return kRtpNoRtp;
} else {
return kRtpDead;
}
}
void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
PayloadUnion* /* specific_payload */,
bool* should_discard_changes) {
*should_discard_changes =
TelephoneEventPayloadType(payload_type) || CNGPayloadType(payload_type);
}
int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
rtc::CritScope cs(&crit_sect_);
assert(num_energy_ <= kRtpCsrcSize);
if (num_energy_ > 0) {
memcpy(array_of_energy, current_remote_energy_,
sizeof(uint8_t) * num_energy_);
}
return num_energy_;
}
int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
RtpFeedback* callback,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const {
if (-1 ==
callback->OnInitializeDecoder(
payload_type, payload_name, specific_payload.Audio.frequency,
specific_payload.Audio.channels, specific_payload.Audio.rate)) {
LOG(LS_ERROR) << "Failed to create decoder for payload type: "
<< payload_name << "/" << static_cast<int>(payload_type);
return -1;
}
return 0;
}
// We are not allowed to have any critsects when calling data_callback.
int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
size_t payload_length,
const AudioPayload& audio_specific,
bool is_red) {
if (payload_length == 0) {
return 0;
}
bool telephone_event_packet =
TelephoneEventPayloadType(rtp_header->header.payloadType);
if (telephone_event_packet) {
rtc::CritScope lock(&crit_sect_);
// RFC 4733 2.3
// 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 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
if (payload_length % 4 != 0) {
return -1;
}
size_t number_of_events = payload_length / 4;
// sanity
if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
}
for (size_t n = 0; n < number_of_events; ++n) {
bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
std::set<uint8_t>::iterator event =
telephone_event_reported_.find(payload_data[4 * n]);
if (event != telephone_event_reported_.end()) {
// we have already seen this event
if (end) {
telephone_event_reported_.erase(payload_data[4 * n]);
}
} else {
if (end) {
// don't add if it's a end of a tone
} else {
telephone_event_reported_.insert(payload_data[4 * n]);
}
}
}
// RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
// should not be a problem since we don't care about the duration
// RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet
}
{
rtc::CritScope lock(&crit_sect_);
// Check if this is a CNG packet, receiver might want to know
if (CNGPayloadType(rtp_header->header.payloadType)) {
rtp_header->type.Audio.isCNG = true;
rtp_header->frameType = kAudioFrameCN;
} else {
rtp_header->frameType = kAudioFrameSpeech;
rtp_header->type.Audio.isCNG = false;
}
// check if it's a DTMF event, hence something we can playout
if (telephone_event_packet) {
if (!telephone_event_forward_to_decoder_) {
// don't forward event to decoder
return 0;
}
std::set<uint8_t>::iterator first =
telephone_event_reported_.begin();
if (first != telephone_event_reported_.end() && *first > 15) {
// don't forward non DTMF events
return 0;
}
}
}
// TODO(holmer): Break this out to have RED parsing handled generically.
if (is_red && !(payload_data[0] & 0x80)) {
// we recive only one frame packed in a RED packet remove the RED wrapper
rtp_header->header.payloadType = payload_data[0];
// only one frame in the RED strip the one byte to help NetEq
return data_callback_->OnReceivedPayloadData(
payload_data + 1, payload_length - 1, rtp_header);
}
rtp_header->type.Audio.channel = audio_specific.channels;
return data_callback_->OnReceivedPayloadData(
payload_data, payload_length, rtp_header);
}
} // namespace webrtc
Reference in New Issue View Git Blame Copy Permalink