| /* |
| * 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_receiver_audio.h" |
| |
| #include <assert.h> // assert |
| #include <math.h> // pow() |
| #include <string.h> // memcpy() |
| |
| #include "common_types.h" // NOLINT(build/include) |
| #include "rtc_base/logging.h" |
| #include "rtc_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) {} |
| |
| RTPReceiverAudio::~RTPReceiverAudio() = default; |
| |
| // 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; |
| } |
| |
| TelephoneEventHandler* RTPReceiverAudio::GetTelephoneEventHandler() { |
| return this; |
| } |
| |
| 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_; |
| } |
| |
| // - 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( |
| int payload_type, |
| const SdpAudioFormat& audio_format) { |
| rtc::CritScope lock(&crit_sect_); |
| |
| if (RtpUtility::StringCompare(audio_format.name.c_str(), "telephone-event", |
| 15)) { |
| telephone_event_payload_type_ = payload_type; |
| } |
| if (RtpUtility::StringCompare(audio_format.name.c_str(), "cn", 2)) { |
| // We support comfort noise at four different frequencies. |
| if (audio_format.clockrate_hz == 8000) { |
| cng_nb_payload_type_ = payload_type; |
| } else if (audio_format.clockrate_hz == 16000) { |
| cng_wb_payload_type_ = payload_type; |
| } else if (audio_format.clockrate_hz == 32000) { |
| cng_swb_payload_type_ = payload_type; |
| } else if (audio_format.clockrate_hz == 48000) { |
| cng_fb_payload_type_ = payload_type; |
| } else { |
| assert(false); |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
| const PayloadUnion& specific_payload, |
| const uint8_t* payload, |
| size_t payload_length, |
| int64_t timestamp_ms) { |
| if (first_packet_received_()) { |
| RTC_LOG(LS_INFO) << "Received first audio RTP packet"; |
| } |
| |
| return ParseAudioCodecSpecific(rtp_header, payload, payload_length, |
| specific_payload.audio_payload()); |
| } |
| |
| 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); |
| } |
| |
| // 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) { |
| RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength); |
| const size_t payload_data_length = |
| payload_length - rtp_header->header.paddingLength; |
| if (payload_data_length == 0) { |
| rtp_header->frameType = kEmptyFrame; |
| return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header); |
| } |
| |
| 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_data_length % 4 != 0) { |
| return -1; |
| } |
| size_t number_of_events = payload_data_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) { |
| RTC_DCHECK_GE(payload_data_length, (4 * n) + 2); |
| 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 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; |
| } |
| } |
| } |
| |
| return data_callback_->OnReceivedPayloadData(payload_data, |
| payload_data_length, rtp_header); |
| } |
| } // namespace webrtc |
