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https://github.com/samsonjs/media.git
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Add support for RTSP Opus
Added Opus RTP packet reader and added support for Opus playback through RTSP Change-Id: Ib6702bd8aafd0bd782e89127ab907061ff06ccb3
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4b72335d51
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be108b9e5e
4 changed files with 166 additions and 0 deletions
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@ -37,6 +37,7 @@ import java.util.Map;
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public final class RtpPayloadFormat {
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public final class RtpPayloadFormat {
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private static final String RTP_MEDIA_AC3 = "AC3";
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private static final String RTP_MEDIA_AC3 = "AC3";
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private static final String RTP_MEDIA_OPUS = "OPUS";
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private static final String RTP_MEDIA_MPEG4_GENERIC = "MPEG4-GENERIC";
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private static final String RTP_MEDIA_MPEG4_GENERIC = "MPEG4-GENERIC";
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private static final String RTP_MEDIA_H264 = "H264";
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private static final String RTP_MEDIA_H264 = "H264";
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private static final String RTP_MEDIA_H265 = "H265";
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private static final String RTP_MEDIA_H265 = "H265";
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@ -45,6 +46,7 @@ public final class RtpPayloadFormat {
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public static boolean isFormatSupported(MediaDescription mediaDescription) {
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public static boolean isFormatSupported(MediaDescription mediaDescription) {
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switch (Ascii.toUpperCase(mediaDescription.rtpMapAttribute.mediaEncoding)) {
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switch (Ascii.toUpperCase(mediaDescription.rtpMapAttribute.mediaEncoding)) {
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case RTP_MEDIA_AC3:
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case RTP_MEDIA_AC3:
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case RTP_MEDIA_OPUS:
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case RTP_MEDIA_H264:
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case RTP_MEDIA_H264:
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case RTP_MEDIA_H265:
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case RTP_MEDIA_H265:
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case RTP_MEDIA_MPEG4_GENERIC:
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case RTP_MEDIA_MPEG4_GENERIC:
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@ -71,6 +73,8 @@ public final class RtpPayloadFormat {
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return MimeTypes.VIDEO_H265;
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return MimeTypes.VIDEO_H265;
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case RTP_MEDIA_MPEG4_GENERIC:
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case RTP_MEDIA_MPEG4_GENERIC:
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return MimeTypes.AUDIO_AAC;
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return MimeTypes.AUDIO_AAC;
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case RTP_MEDIA_OPUS:
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return MimeTypes.AUDIO_OPUS;
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default:
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default:
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throw new IllegalArgumentException(mediaType);
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throw new IllegalArgumentException(mediaType);
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}
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}
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@ -121,6 +121,14 @@ import com.google.common.collect.ImmutableMap;
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checkArgument(!fmtpParameters.isEmpty());
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checkArgument(!fmtpParameters.isEmpty());
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processAacFmtpAttribute(formatBuilder, fmtpParameters, channelCount, clockRate);
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processAacFmtpAttribute(formatBuilder, fmtpParameters, channelCount, clockRate);
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break;
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break;
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case MimeTypes.AUDIO_OPUS:
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// RFC7587 Section 7
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checkArgument(channelCount == 2, "Invalid channel count");
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// RFC7587 Section 6.1
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// the RTP timestamp is incremented with a 48000 Hz clock rate
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// for all modes of Opus and all sampling rates.
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checkArgument(clockRate == 48000, "Invalid sampling rate");
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break;
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case MimeTypes.VIDEO_H264:
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case MimeTypes.VIDEO_H264:
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checkArgument(!fmtpParameters.isEmpty());
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checkArgument(!fmtpParameters.isEmpty());
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processH264FmtpAttribute(formatBuilder, fmtpParameters);
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processH264FmtpAttribute(formatBuilder, fmtpParameters);
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@ -36,6 +36,8 @@ import androidx.media3.exoplayer.rtsp.RtpPayloadFormat;
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return new RtpAc3Reader(payloadFormat);
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return new RtpAc3Reader(payloadFormat);
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case MimeTypes.AUDIO_AAC:
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case MimeTypes.AUDIO_AAC:
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return new RtpAacReader(payloadFormat);
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return new RtpAacReader(payloadFormat);
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case MimeTypes.AUDIO_OPUS:
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return new RtpOpusReader(payloadFormat);
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case MimeTypes.VIDEO_H264:
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case MimeTypes.VIDEO_H264:
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return new RtpH264Reader(payloadFormat);
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return new RtpH264Reader(payloadFormat);
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case MimeTypes.VIDEO_H265:
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case MimeTypes.VIDEO_H265:
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@ -0,0 +1,152 @@
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/*
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* Copyright 2022 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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package androidx.media3.exoplayer.rtsp.reader;
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import static androidx.media3.common.util.Assertions.checkArgument;
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import static androidx.media3.common.util.Assertions.checkStateNotNull;
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import androidx.media3.common.C;
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import androidx.media3.common.Format;
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import androidx.media3.common.util.Log;
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import androidx.media3.common.util.ParsableByteArray;
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import androidx.media3.common.util.Util;
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import androidx.media3.exoplayer.rtsp.RtpPacket;
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import androidx.media3.exoplayer.rtsp.RtpPayloadFormat;
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import androidx.media3.extractor.ExtractorOutput;
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import androidx.media3.extractor.OpusUtil;
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import androidx.media3.extractor.TrackOutput;
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import java.util.List;
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import org.checkerframework.checker.nullness.qual.MonotonicNonNull;
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/**
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* Parses an OPUS byte stream carried on RTP packets and extracts individual samples. Refer to
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* RFC7845 for more details.
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*/
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/* package */ final class RtpOpusReader implements RtpPayloadReader {
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private static final String TAG = "RtpOpusReader";
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private final RtpPayloadFormat payloadFormat;
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private @MonotonicNonNull TrackOutput trackOutput;
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private long firstReceivedTimestamp;
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private long startTimeOffsetUs;
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private final int sampleRate;
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private int previousSequenceNumber;
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private boolean foundOpusIDHeader;
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private boolean foundOpusCommentHeader;
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public RtpOpusReader(RtpPayloadFormat payloadFormat) {
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this.payloadFormat = payloadFormat;
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this.firstReceivedTimestamp = C.INDEX_UNSET;
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this.sampleRate = this.payloadFormat.clockRate;
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this.previousSequenceNumber = C.INDEX_UNSET;
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this.foundOpusIDHeader = false;
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this.foundOpusCommentHeader = false;
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}
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// RtpPayloadReader implementation.
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@Override
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public void createTracks(ExtractorOutput extractorOutput, int trackId) {
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trackOutput = extractorOutput.track(trackId, C.TRACK_TYPE_AUDIO);
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trackOutput.format(payloadFormat.format);
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}
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@Override
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public void onReceivingFirstPacket(long timestamp, int sequenceNumber) {
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this.firstReceivedTimestamp = timestamp;
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}
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@Override
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public void consume(
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ParsableByteArray data, long timestamp, int sequenceNumber, boolean rtpMarker) {
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checkStateNotNull(trackOutput);
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/* RFC7845 Section 3
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* +---------+ +----------------+ +--------------------+ +-----
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* |ID Header| | Comment Header | |Audio Data Packet 1 | | ...
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* +---------+ +----------------+ +--------------------+ +-----
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*/
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if (!foundOpusIDHeader) {
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int currPosition = data.getPosition();
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checkArgument(isOpusIDHeader(data), "ID Header missing");
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data.setPosition(currPosition);
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List<byte[]> initializationData = OpusUtil.buildInitializationData(data.getData());
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Format.Builder formatBuilder = payloadFormat.format.buildUpon();
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formatBuilder.setInitializationData(initializationData);
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trackOutput.format(formatBuilder.build());
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foundOpusIDHeader = true;
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} else if (!foundOpusCommentHeader) {
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// Comment Header RFC7845 Section 5.2
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String header = data.readString(8);
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checkArgument(header.equals("OpusTags"), "Comment Header should follow ID Header");
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foundOpusCommentHeader = true;
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} else {
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// Check that this packet is in the sequence of the previous packet.
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int expectedSequenceNumber = RtpPacket.getNextSequenceNumber(previousSequenceNumber);
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if (sequenceNumber != expectedSequenceNumber) {
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Log.w(
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TAG,
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Util.formatInvariant(
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"Received RTP packet with unexpected sequence number. Expected: %d; received: %d.",
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expectedSequenceNumber, sequenceNumber));
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}
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// sending opus data
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int size = data.bytesLeft();
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trackOutput.sampleData(data, size);
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long timeUs =
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toSampleTimeUs(startTimeOffsetUs, timestamp, firstReceivedTimestamp, sampleRate);
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trackOutput.sampleMetadata(timeUs, C.BUFFER_FLAG_KEY_FRAME, size, 0, null);
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}
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previousSequenceNumber = sequenceNumber;
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}
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@Override
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public void seek(long nextRtpTimestamp, long timeUs) {
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firstReceivedTimestamp = nextRtpTimestamp;
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startTimeOffsetUs = timeUs;
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}
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// Internal methods.
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private static boolean isOpusIDHeader(ParsableByteArray data) {
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int sampleSize = data.limit();
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String header = data.readString(8);
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// Identification header RFC7845 Section 5.1
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if (sampleSize < 19 || !header.equals("OpusHead")) {
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Log.e(
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TAG,
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Util.formatInvariant(
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"first data octet of the RTP packet is not the beginning of a OpusHeader "
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+ "Dropping current packet"));
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return false;
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}
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checkArgument(data.readUnsignedByte() == 1, "version number must always be 1");
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return true;
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}
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/** Returns the correct sample time from RTP timestamp, accounting for the OPUS sampling rate. */
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private static long toSampleTimeUs(
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long startTimeOffsetUs, long rtpTimestamp, long firstReceivedRtpTimestamp, int sampleRate) {
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return startTimeOffsetUs
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+ Util.scaleLargeTimestamp(
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rtpTimestamp - firstReceivedRtpTimestamp,
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/* multiplier= */ C.MICROS_PER_SECOND,
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/* divisor= */ sampleRate);
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}
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}
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