| /* |
| * 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) {} |
| |
| RTPReceiverAudio::~RTPReceiverAudio() = default; |
| |
| // - 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::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()); |
| } |
| |
| // 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); |
| } |
| |
| return data_callback_->OnReceivedPayloadData(payload_data, |
| payload_data_length, rtp_header); |
| } |
| } // namespace webrtc |
