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webrtc: support reading G711 16khz tracks (#2848)
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@aler9
aler9 committed Jun 9, 2024
1 parent 427fea3 commit 4c8d0be
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Showing 4 changed files with 194 additions and 50 deletions.
65 changes: 43 additions & 22 deletions internal/protocols/webrtc/outgoing_track.go
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Original file line number Diff line number Diff line change
Expand Up @@ -92,59 +92,80 @@ func (t *OutgoingTrack) codecParameters() (webrtc.RTPCodecParameters, error) {
}, nil

case *format.G711:
if forma.SampleRate != 8000 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported G711 sample rate")
// These are the sample rates and channels supported by Chrome.
// Different sample rates and channels can be streamed too but we don't want compatibility issues.
// https://webrtc.googlesource.com/src/+/refs/heads/main/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc#23
if forma.ClockRate() != 8000 && forma.ClockRate() != 16000 &&
forma.ClockRate() != 32000 && forma.ClockRate() != 48000 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported clock rate: %d", forma.ClockRate())
}

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if forma.ChannelCount != 1 && forma.ChannelCount != 2 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported channel count: %d", forma.ChannelCount)

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}

if forma.MULaw {
if forma.ChannelCount != 1 {
if forma.SampleRate == 8000 {
if forma.MULaw {
if forma.ChannelCount != 1 {
return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: uint32(forma.SampleRate),
Channels: uint16(forma.ChannelCount),
},
PayloadType: 96,
}, nil
}

return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: 8000,
},
PayloadType: 0,
}, nil
}

if forma.ChannelCount != 1 {
return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: uint32(forma.SampleRate),
Channels: uint16(forma.ChannelCount),
},
PayloadType: 96,
}, nil
}

return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMU,
ClockRate: 8000,
},
PayloadType: 0,
}, nil
}

if forma.ChannelCount != 1 {
return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: uint32(forma.SampleRate),
Channels: uint16(forma.ChannelCount),
ClockRate: 8000,
},
PayloadType: 96,
PayloadType: 8,
}, nil
}

return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypePCMA,
ClockRate: 8000,
MimeType: mimeTypeL16,
ClockRate: uint32(forma.ClockRate()),
Channels: uint16(forma.ChannelCount),
},
PayloadType: 8,
PayloadType: 96,
}, nil

case *format.LPCM:
if forma.BitDepth != 16 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported LPCM bit depth: %d", forma.BitDepth)
}

if forma.ClockRate() != 8000 && forma.ClockRate() != 16000 && forma.ClockRate() != 48000 {
// These are the sample rates and channels supported by Chrome.
// Different sample rates and channels can be streamed too but we don't want compatibility issues.
// https://webrtc.googlesource.com/src/+/refs/heads/main/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc#23
if forma.ClockRate() != 8000 && forma.ClockRate() != 16000 &&
forma.ClockRate() != 32000 && forma.ClockRate() != 48000 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported clock rate: %d", forma.ClockRate())
}

if forma.ChannelCount != 1 && forma.ChannelCount != 2 {
return webrtc.RTPCodecParameters{}, fmt.Errorf("unsupported channel count: %d", forma.ChannelCount)
}
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71 changes: 50 additions & 21 deletions internal/protocols/webrtc/peer_connection_test.go
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Original file line number Diff line number Diff line change
Expand Up @@ -114,7 +114,35 @@ func TestPeerConnectionPublishRead(t *testing.T) {
&format.G722{},
},
{
"g711 pcma stereo",
"g711 pcma 8khz mono",
&format.G711{
PayloadTyp: 8,
SampleRate: 8000,
ChannelCount: 1,
},
&format.G711{
PayloadTyp: 8,
SampleRate: 8000,
ChannelCount: 1,
},
},
{
"g711 pcmu 8khz mono",
&format.G711{
MULaw: true,
PayloadTyp: 0,
SampleRate: 8000,
ChannelCount: 1,
},
&format.G711{
MULaw: true,
PayloadTyp: 0,
SampleRate: 8000,
ChannelCount: 1,
},
},
{
"g711 pcma 8khz stereo",
&format.G711{
PayloadTyp: 96,
SampleRate: 8000,
Expand All @@ -127,7 +155,7 @@ func TestPeerConnectionPublishRead(t *testing.T) {
},
},
{
"g711 pcmu stereo",
"g711 pcmu 8khz stereo",
&format.G711{
MULaw: true,
PayloadTyp: 96,
Expand All @@ -142,35 +170,36 @@ func TestPeerConnectionPublishRead(t *testing.T) {
},
},
{
"g711 pcma mono",
"g711 pcma 16khz stereo",
&format.G711{
PayloadTyp: 8,
SampleRate: 8000,
ChannelCount: 1,
PayloadTyp: 96,
SampleRate: 16000,
ChannelCount: 2,
},
&format.G711{
PayloadTyp: 8,
SampleRate: 8000,
ChannelCount: 1,
&format.LPCM{
PayloadTyp: 96,
BitDepth: 16,
SampleRate: 16000,
ChannelCount: 2,
},
},
{
"g711 pcmu mono",
"g711 pcmu 16khz stereo",
&format.G711{
MULaw: true,
PayloadTyp: 0,
SampleRate: 8000,
ChannelCount: 1,
PayloadTyp: 96,
SampleRate: 16000,
ChannelCount: 2,
},
&format.G711{
MULaw: true,
PayloadTyp: 0,
SampleRate: 8000,
ChannelCount: 1,
&format.LPCM{
PayloadTyp: 96,
BitDepth: 16,
SampleRate: 16000,
ChannelCount: 2,
},
},
{
"l16 8000 stereo",
"l16 8khz stereo",
&format.LPCM{
PayloadTyp: 96,
BitDepth: 16,
Expand All @@ -185,7 +214,7 @@ func TestPeerConnectionPublishRead(t *testing.T) {
},
},
{
"l16 16000 stereo",
"l16 16khz stereo",
&format.LPCM{
PayloadTyp: 96,
BitDepth: 16,
Expand Down
17 changes: 16 additions & 1 deletion internal/servers/webrtc/server_test.go
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Original file line number Diff line number Diff line change
Expand Up @@ -431,7 +431,7 @@ func TestServerRead(t *testing.T) {
[]byte{1, 2},
},*/
{
"g711",
"g711 8khz mono",
[]*description.Media{{
Type: description.MediaTypeAudio,
Formats: []format.Format{&format.G711{
Expand All @@ -445,6 +445,21 @@ func TestServerRead(t *testing.T) {
},
[]byte{1, 2, 3},
},
{
"g711 16khz stereo",
[]*description.Media{{
Type: description.MediaTypeAudio,
Formats: []format.Format{&format.G711{
MULaw: true,
SampleRate: 16000,
ChannelCount: 2,
}},
}},
&unit.G711{
Samples: []byte{1, 2, 3, 4},
},
[]byte{0x86, 0x84, 0x8a, 0x84, 0x8e, 0x84, 0x92, 0x84},
},
{
"lpcm",
[]*description.Media{{
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91 changes: 85 additions & 6 deletions internal/servers/webrtc/session.go
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Original file line number Diff line number Diff line change
Expand Up @@ -2,6 +2,7 @@ package webrtc

import (
"context"
"crypto/rand"
"encoding/hex"
"errors"
"fmt"
Expand All @@ -18,6 +19,7 @@ import (
"github.com/bluenviron/gortsplib/v4/pkg/format/rtpvp8"
"github.com/bluenviron/gortsplib/v4/pkg/format/rtpvp9"
"github.com/bluenviron/gortsplib/v4/pkg/rtptime"
"github.com/bluenviron/mediacommon/pkg/codecs/g711"
"github.com/google/uuid"
"github.com/pion/ice/v2"
"github.com/pion/sdp/v3"
Expand All @@ -43,6 +45,15 @@ func uint16Ptr(v uint16) *uint16 {
return &v
}

func randUint32() (uint32, error) {
var b [4]byte
_, err := rand.Read(b[:])
if err != nil {
return 0, err
}

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return uint32(b[0])<<24 | uint32(b[1])<<16 | uint32(b[2])<<8 | uint32(b[3]), nil
}

func findVideoTrack(
stream *stream.Stream,
writer *asyncwriter.Writer,
Expand Down Expand Up @@ -254,13 +265,72 @@ func findAudioTrack(

if g711Format != nil {
return g711Format, func(track *webrtc.OutgoingTrack) error {
stream.AddReader(writer, media, g711Format, func(u unit.Unit) error {
for _, pkt := range u.GetRTPPackets() {
track.WriteRTP(pkt) //nolint:errcheck
if g711Format.SampleRate == 8000 {
curTimestamp, err := randUint32()
if err != nil {
return err

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}

return nil
})
stream.AddReader(writer, media, g711Format, func(u unit.Unit) error {
for _, pkt := range u.GetRTPPackets() {
// recompute timestamp from scratch.
// Chrome requires a precise timestamp that FFmpeg doesn't provide.
pkt.Timestamp = curTimestamp
curTimestamp += uint32(len(pkt.Payload)) / uint32(g711Format.ChannelCount)

track.WriteRTP(pkt) //nolint:errcheck
}

return nil
})
} else {
encoder := &rtplpcm.Encoder{
PayloadType: 96,
PayloadMaxSize: webrtcPayloadMaxSize,
BitDepth: 16,
ChannelCount: g711Format.ChannelCount,
}
err := encoder.Init()
if err != nil {
return err
}

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curTimestamp, err := randUint32()
if err != nil {
return err
}

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stream.AddReader(writer, media, g711Format, func(u unit.Unit) error {
tunit := u.(*unit.G711)

if tunit.Samples == nil {
return nil
}

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var lpcmSamples []byte
if g711Format.MULaw {
lpcmSamples = g711.DecodeMulaw(tunit.Samples)
} else {
lpcmSamples = g711.DecodeAlaw(tunit.Samples)
}

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packets, err := encoder.Encode(lpcmSamples)
if err != nil {
return nil //nolint:nilerr
}

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for _, pkt := range packets {
// recompute timestamp from scratch.
// Chrome requires a precise timestamp that FFmpeg doesn't provide.
pkt.Timestamp = curTimestamp
curTimestamp += uint32(len(pkt.Payload)) / 2 / uint32(g711Format.ChannelCount)

track.WriteRTP(pkt) //nolint:errcheck
}

return nil
})
}
return nil
}
}
Expand All @@ -281,6 +351,11 @@ func findAudioTrack(
return err
}

curTimestamp, err := randUint32()
if err != nil {
return err
}

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stream.AddReader(writer, media, lpcmFormat, func(u unit.Unit) error {
tunit := u.(*unit.LPCM)

Expand All @@ -294,7 +369,11 @@ func findAudioTrack(
}

for _, pkt := range packets {
pkt.Timestamp += tunit.RTPPackets[0].Timestamp
// recompute timestamp from scratch.
// Chrome requires a precise timestamp that FFmpeg doesn't provide.
pkt.Timestamp = curTimestamp
curTimestamp += uint32(len(pkt.Payload)) / 2 / uint32(lpcmFormat.ChannelCount)

track.WriteRTP(pkt) //nolint:errcheck
}

Expand Down

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