mirror of
https://github.com/oxen-io/session-android.git
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248 lines
7.5 KiB
C++
248 lines
7.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "RTPFile.h"
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#include <stdlib.h>
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#ifdef WIN32
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# include <Winsock2.h>
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#else
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# include <arpa/inet.h>
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#endif
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#include "audio_coding_module.h"
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#include "engine_configurations.h"
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#include "gtest/gtest.h" // TODO (tlegrand): Consider removing usage of gtest.
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#include "rw_lock_wrapper.h"
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namespace webrtc {
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void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
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const uint8_t* rtpHeader) {
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rtpInfo->header.payloadType = rtpHeader[1];
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rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
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rtpHeader[3];
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rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
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(static_cast<uint32_t>(rtpHeader[5]) << 16) |
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(static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
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rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
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(static_cast<uint32_t>(rtpHeader[9]) << 16) |
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(static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11];
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}
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void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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int16_t seqNo, uint32_t timeStamp,
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uint32_t ssrc) {
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rtpHeader[0] = (unsigned char) 0x80;
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rtpHeader[1] = (unsigned char) (payloadType & 0xFF);
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rtpHeader[2] = (unsigned char) ((seqNo >> 8) & 0xFF);
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rtpHeader[3] = (unsigned char) ((seqNo) & 0xFF);
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rtpHeader[4] = (unsigned char) ((timeStamp >> 24) & 0xFF);
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rtpHeader[5] = (unsigned char) ((timeStamp >> 16) & 0xFF);
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rtpHeader[6] = (unsigned char) ((timeStamp >> 8) & 0xFF);
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rtpHeader[7] = (unsigned char) (timeStamp & 0xFF);
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rtpHeader[8] = (unsigned char) ((ssrc >> 24) & 0xFF);
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rtpHeader[9] = (unsigned char) ((ssrc >> 16) & 0xFF);
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rtpHeader[10] = (unsigned char) ((ssrc >> 8) & 0xFF);
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rtpHeader[11] = (unsigned char) (ssrc & 0xFF);
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}
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RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
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const uint8_t* payloadData, uint16_t payloadSize,
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uint32_t frequency)
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: payloadType(payloadType),
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timeStamp(timeStamp),
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seqNo(seqNo),
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payloadSize(payloadSize),
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frequency(frequency) {
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if (payloadSize > 0) {
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this->payloadData = new uint8_t[payloadSize];
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memcpy(this->payloadData, payloadData, payloadSize);
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}
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}
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RTPPacket::~RTPPacket() {
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delete[] payloadData;
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}
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RTPBuffer::RTPBuffer() {
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_queueRWLock = RWLockWrapper::CreateRWLock();
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}
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RTPBuffer::~RTPBuffer() {
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delete _queueRWLock;
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}
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void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const uint16_t payloadSize, uint32_t frequency) {
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RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
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payloadSize, frequency);
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_queueRWLock->AcquireLockExclusive();
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_rtpQueue.push(packet);
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_queueRWLock->ReleaseLockExclusive();
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}
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uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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uint16_t payloadSize, uint32_t* offset) {
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_queueRWLock->AcquireLockShared();
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RTPPacket *packet = _rtpQueue.front();
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_rtpQueue.pop();
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_queueRWLock->ReleaseLockShared();
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rtpInfo->header.markerBit = 1;
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rtpInfo->header.payloadType = packet->payloadType;
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rtpInfo->header.sequenceNumber = packet->seqNo;
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rtpInfo->header.ssrc = 0;
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rtpInfo->header.timestamp = packet->timeStamp;
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if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
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memcpy(payloadData, packet->payloadData, packet->payloadSize);
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} else {
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return 0;
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}
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*offset = (packet->timeStamp / (packet->frequency / 1000));
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return packet->payloadSize;
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}
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bool RTPBuffer::EndOfFile() const {
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_queueRWLock->AcquireLockShared();
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bool eof = _rtpQueue.empty();
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_queueRWLock->ReleaseLockShared();
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return eof;
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}
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void RTPFile::Open(const char *filename, const char *mode) {
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if ((_rtpFile = fopen(filename, mode)) == NULL) {
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printf("Cannot write file %s.\n", filename);
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ADD_FAILURE() << "Unable to write file";
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exit(1);
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}
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}
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void RTPFile::Close() {
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if (_rtpFile != NULL) {
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fclose(_rtpFile);
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_rtpFile = NULL;
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}
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}
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void RTPFile::WriteHeader() {
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// Write data in a format that NetEQ and RTP Play can parse
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fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
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uint32_t dummy_variable = 0;
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// should be converted to network endian format, but does not matter when 0
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if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
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return;
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}
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fflush(_rtpFile);
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}
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void RTPFile::ReadHeader() {
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uint32_t start_sec, start_usec, source;
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uint16_t port, padding;
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char fileHeader[40];
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EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
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EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
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start_sec = ntohl(start_sec);
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EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
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start_usec = ntohl(start_usec);
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EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
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source = ntohl(source);
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EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
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port = ntohs(port);
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EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
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padding = ntohs(padding);
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}
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void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const uint16_t payloadSize, uint32_t frequency) {
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/* write RTP packet to file */
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uint8_t rtpHeader[12];
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MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
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uint16_t lengthBytes = htons(12 + payloadSize + 8);
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uint16_t plen = htons(12 + payloadSize);
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uint32_t offsetMs;
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offsetMs = (timeStamp / (frequency / 1000));
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offsetMs = htonl(offsetMs);
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if (fwrite(&lengthBytes, 2, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&plen, 2, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(&offsetMs, 4, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(rtpHeader, 12, 1, _rtpFile) != 1) {
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return;
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}
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if (fwrite(payloadData, 1, payloadSize, _rtpFile) != payloadSize) {
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return;
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}
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}
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uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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uint16_t payloadSize, uint32_t* offset) {
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uint16_t lengthBytes;
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uint16_t plen;
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uint8_t rtpHeader[12];
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size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
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/* Check if we have reached end of file. */
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if ((read_len == 0) && feof(_rtpFile)) {
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_rtpEOF = true;
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return 0;
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}
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EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
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EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
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lengthBytes = ntohs(lengthBytes);
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plen = ntohs(plen);
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*offset = ntohl(*offset);
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EXPECT_GT(plen, 11);
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EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
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ParseRTPHeader(rtpInfo, rtpHeader);
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rtpInfo->type.Audio.isCNG = false;
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rtpInfo->type.Audio.channel = 1;
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EXPECT_EQ(lengthBytes, plen + 8);
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if (plen == 0) {
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return 0;
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}
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if (payloadSize < (lengthBytes - 20)) {
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return 0;
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}
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if (lengthBytes < 20) {
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return 0;
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}
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lengthBytes -= 20;
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EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
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return lengthBytes;
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}
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} // namespace webrtc
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