Add support for RTSP Opus

Added Opus RTP packet reader and added support for Opus
playback through RTSP

Change-Id: Ib6702bd8aafd0bd782e89127ab907061ff06ccb3
This commit is contained in:
Manisha Jajoo 2022-03-03 18:57:38 +05:30
parent 4b72335d51
commit be108b9e5e
4 changed files with 166 additions and 0 deletions

View file

@ -37,6 +37,7 @@ import java.util.Map;
public final class RtpPayloadFormat {
private static final String RTP_MEDIA_AC3 = "AC3";
private static final String RTP_MEDIA_OPUS = "OPUS";
private static final String RTP_MEDIA_MPEG4_GENERIC = "MPEG4-GENERIC";
private static final String RTP_MEDIA_H264 = "H264";
private static final String RTP_MEDIA_H265 = "H265";
@ -45,6 +46,7 @@ public final class RtpPayloadFormat {
public static boolean isFormatSupported(MediaDescription mediaDescription) {
switch (Ascii.toUpperCase(mediaDescription.rtpMapAttribute.mediaEncoding)) {
case RTP_MEDIA_AC3:
case RTP_MEDIA_OPUS:
case RTP_MEDIA_H264:
case RTP_MEDIA_H265:
case RTP_MEDIA_MPEG4_GENERIC:
@ -71,6 +73,8 @@ public final class RtpPayloadFormat {
return MimeTypes.VIDEO_H265;
case RTP_MEDIA_MPEG4_GENERIC:
return MimeTypes.AUDIO_AAC;
case RTP_MEDIA_OPUS:
return MimeTypes.AUDIO_OPUS;
default:
throw new IllegalArgumentException(mediaType);
}

View file

@ -121,6 +121,14 @@ import com.google.common.collect.ImmutableMap;
checkArgument(!fmtpParameters.isEmpty());
processAacFmtpAttribute(formatBuilder, fmtpParameters, channelCount, clockRate);
break;
case MimeTypes.AUDIO_OPUS:
// RFC7587 Section 7
checkArgument(channelCount == 2, "Invalid channel count");
// RFC7587 Section 6.1
// the RTP timestamp is incremented with a 48000 Hz clock rate
// for all modes of Opus and all sampling rates.
checkArgument(clockRate == 48000, "Invalid sampling rate");
break;
case MimeTypes.VIDEO_H264:
checkArgument(!fmtpParameters.isEmpty());
processH264FmtpAttribute(formatBuilder, fmtpParameters);

View file

@ -36,6 +36,8 @@ import androidx.media3.exoplayer.rtsp.RtpPayloadFormat;
return new RtpAc3Reader(payloadFormat);
case MimeTypes.AUDIO_AAC:
return new RtpAacReader(payloadFormat);
case MimeTypes.AUDIO_OPUS:
return new RtpOpusReader(payloadFormat);
case MimeTypes.VIDEO_H264:
return new RtpH264Reader(payloadFormat);
case MimeTypes.VIDEO_H265:

View file

@ -0,0 +1,152 @@
/*
* Copyright 2022 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
package androidx.media3.exoplayer.rtsp.reader;
import static androidx.media3.common.util.Assertions.checkArgument;
import static androidx.media3.common.util.Assertions.checkStateNotNull;
import androidx.media3.common.C;
import androidx.media3.common.Format;
import androidx.media3.common.util.Log;
import androidx.media3.common.util.ParsableByteArray;
import androidx.media3.common.util.Util;
import androidx.media3.exoplayer.rtsp.RtpPacket;
import androidx.media3.exoplayer.rtsp.RtpPayloadFormat;
import androidx.media3.extractor.ExtractorOutput;
import androidx.media3.extractor.OpusUtil;
import androidx.media3.extractor.TrackOutput;
import java.util.List;
import org.checkerframework.checker.nullness.qual.MonotonicNonNull;
/**
* Parses an OPUS byte stream carried on RTP packets and extracts individual samples. Refer to
* RFC7845 for more details.
*/
/* package */ final class RtpOpusReader implements RtpPayloadReader {
private static final String TAG = "RtpOpusReader";
private final RtpPayloadFormat payloadFormat;
private @MonotonicNonNull TrackOutput trackOutput;
private long firstReceivedTimestamp;
private long startTimeOffsetUs;
private final int sampleRate;
private int previousSequenceNumber;
private boolean foundOpusIDHeader;
private boolean foundOpusCommentHeader;
public RtpOpusReader(RtpPayloadFormat payloadFormat) {
this.payloadFormat = payloadFormat;
this.firstReceivedTimestamp = C.INDEX_UNSET;
this.sampleRate = this.payloadFormat.clockRate;
this.previousSequenceNumber = C.INDEX_UNSET;
this.foundOpusIDHeader = false;
this.foundOpusCommentHeader = false;
}
// RtpPayloadReader implementation.
@Override
public void createTracks(ExtractorOutput extractorOutput, int trackId) {
trackOutput = extractorOutput.track(trackId, C.TRACK_TYPE_AUDIO);
trackOutput.format(payloadFormat.format);
}
@Override
public void onReceivingFirstPacket(long timestamp, int sequenceNumber) {
this.firstReceivedTimestamp = timestamp;
}
@Override
public void consume(
ParsableByteArray data, long timestamp, int sequenceNumber, boolean rtpMarker) {
checkStateNotNull(trackOutput);
/* RFC7845 Section 3
* +---------+ +----------------+ +--------------------+ +-----
* |ID Header| | Comment Header | |Audio Data Packet 1 | | ...
* +---------+ +----------------+ +--------------------+ +-----
*/
if (!foundOpusIDHeader) {
int currPosition = data.getPosition();
checkArgument(isOpusIDHeader(data), "ID Header missing");
data.setPosition(currPosition);
List<byte[]> initializationData = OpusUtil.buildInitializationData(data.getData());
Format.Builder formatBuilder = payloadFormat.format.buildUpon();
formatBuilder.setInitializationData(initializationData);
trackOutput.format(formatBuilder.build());
foundOpusIDHeader = true;
} else if (!foundOpusCommentHeader) {
// Comment Header RFC7845 Section 5.2
String header = data.readString(8);
checkArgument(header.equals("OpusTags"), "Comment Header should follow ID Header");
foundOpusCommentHeader = true;
} else {
// Check that this packet is in the sequence of the previous packet.
int expectedSequenceNumber = RtpPacket.getNextSequenceNumber(previousSequenceNumber);
if (sequenceNumber != expectedSequenceNumber) {
Log.w(
TAG,
Util.formatInvariant(
"Received RTP packet with unexpected sequence number. Expected: %d; received: %d.",
expectedSequenceNumber, sequenceNumber));
}
// sending opus data
int size = data.bytesLeft();
trackOutput.sampleData(data, size);
long timeUs =
toSampleTimeUs(startTimeOffsetUs, timestamp, firstReceivedTimestamp, sampleRate);
trackOutput.sampleMetadata(timeUs, C.BUFFER_FLAG_KEY_FRAME, size, 0, null);
}
previousSequenceNumber = sequenceNumber;
}
@Override
public void seek(long nextRtpTimestamp, long timeUs) {
firstReceivedTimestamp = nextRtpTimestamp;
startTimeOffsetUs = timeUs;
}
// Internal methods.
private static boolean isOpusIDHeader(ParsableByteArray data) {
int sampleSize = data.limit();
String header = data.readString(8);
// Identification header RFC7845 Section 5.1
if (sampleSize < 19 || !header.equals("OpusHead")) {
Log.e(
TAG,
Util.formatInvariant(
"first data octet of the RTP packet is not the beginning of a OpusHeader "
+ "Dropping current packet"));
return false;
}
checkArgument(data.readUnsignedByte() == 1, "version number must always be 1");
return true;
}
/** Returns the correct sample time from RTP timestamp, accounting for the OPUS sampling rate. */
private static long toSampleTimeUs(
long startTimeOffsetUs, long rtpTimestamp, long firstReceivedRtpTimestamp, int sampleRate) {
return startTimeOffsetUs
+ Util.scaleLargeTimestamp(
rtpTimestamp - firstReceivedRtpTimestamp,
/* multiplier= */ C.MICROS_PER_SECOND,
/* divisor= */ sampleRate);
}
}