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Adding Opus stereo support to WebRTC

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This CL adds support for sending and receiving stereo using the Opus codec.

BUG=issue1013

Review URL: https://webrtc-codereview.appspot.com/930008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2012-11-07 08:07:29 +00:00
parent 6dddfe9c35
commit 0ad3c1af0a
14 changed files with 276 additions and 36 deletions
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25
webrtc/modules/audio_coding/main/source/acm_codec_database.cc
View File

@ -107,10 +107,10 @@ namespace webrtc {
// codecs. Note! There are a limited number of payload types. If more codecs
// are defined they will receive reserved fixed payload types (values 69-95).
const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
105, 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 121,
92, 91, 90, 89, 88, 87, 86, 85, 84, 83, 82, 81,
80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69,
68, 67
105, 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 92,
91, 90, 89, 88, 87, 86, 85, 84, 83, 82, 81, 80,
79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
67, 66
};
// Creates database with all supported codecs at compile time.
@ -189,8 +189,11 @@ const CodecInst ACMCodecDB::database_[] = {
{3, "GSM", 8000, 160, 1, 13200},
#endif
#ifdef WEBRTC_CODEC_OPUS
// Opus supports 48, 24, 16, 12, 8 kHz.
// Opus internally supports 48, 24, 16, 12, 8 kHz.
// Mono
{120, "opus", 48000, 960, 1, 32000},
// Stereo
{121, "opus", 48000, 960, 2, 32000},
#endif
#ifdef WEBRTC_CODEC_SPEEX
{kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
@ -282,6 +285,9 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
#ifdef WEBRTC_CODEC_OPUS
// Opus supports frames shorter than 10ms,
// but it doesn't help us to use them.
// Mono
{1, {960}, 0, 2},
// Stereo
{1, {960}, 0, 2},
#endif
#ifdef WEBRTC_CODEC_SPEEX
@ -369,7 +375,10 @@ const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
kDecoderGSMFR,
#endif
#ifdef WEBRTC_CODEC_OPUS
// Mono
kDecoderOpus,
// Stereo
kDecoderOpus_2ch,
#endif
#ifdef WEBRTC_CODEC_SPEEX
kDecoderSPEEX_8,
@ -758,7 +767,11 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
#endif
} else if (!STR_CASE_CMP(codec_inst->plname, "opus")) {
#ifdef WEBRTC_CODEC_OPUS
return new ACMOpus(kOpus);
if (codec_inst->channels == 1) {
return new ACMOpus(kOpus);
} else {
return new ACMOpus(kOpus_2ch);
}
#endif
} else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
#ifdef WEBRTC_CODEC_SPEEX

6
webrtc/modules/audio_coding/main/source/acm_codec_database.h
View File

@ -92,7 +92,10 @@ class ACMCodecDB {
, kGSMFR
#endif
#ifdef WEBRTC_CODEC_OPUS
// Mono
, kOpus
// Stereo
, kOpus_2ch
#endif
#ifdef WEBRTC_CODEC_SPEEX
, kSPEEX8
@ -175,7 +178,10 @@ class ACMCodecDB {
enum {kSPEEX16 = -1};
#endif
#ifndef WEBRTC_CODEC_OPUS
// Mono
enum {kOpus = -1};
// Stereo
enum {kOpus_2ch = -1};
#endif
#ifndef WEBRTC_CODEC_AVT
enum {kAVT = -1};

70
webrtc/modules/audio_coding/main/source/acm_opus.cc
View File

@ -29,7 +29,8 @@ ACMOpus::ACMOpus(int16_t /* codecID */)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL),
_sampleFreq(0),
_bitrate(0) {
_bitrate(0),
_channels(1) {
return;
}
@ -91,18 +92,26 @@ int16_t ACMOpus::SetBitRateSafe(const int32_t /*rate*/) {
return -1;
}
bool ACMOpus::IsTrueStereoCodec() {
return true;
}
void ACMOpus::SplitStereoPacket(uint8_t* /*payload*/,
int32_t* /*payload_length*/) {}
#else //===================== Actual Implementation =======================
ACMOpus::ACMOpus(int16_t codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL),
_sampleFreq(32000), // Default sampling frequency.
_bitrate(20000) { // Default bit-rate.
_bitrate(20000), // Default bit-rate.
_channels(1) { // Default mono
_codecID = codecID;
// Opus has internal DTX, but we dont use it for now.
_hasInternalDTX = false;
if (_codecID != ACMCodecDB::kOpus) {
if ((_codecID != ACMCodecDB::kOpus) && (_codecID != ACMCodecDB::kOpus_2ch)) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Wrong codec id for Opus.");
_sampleFreq = -1;
@ -140,7 +149,7 @@ int16_t ACMOpus::InternalEncode(uint8_t* bitStream, int16_t* bitStreamLenByte) {
// Increment the read index. This tells the caller how far
// we have gone forward in reading the audio buffer.
_inAudioIxRead += _frameLenSmpl;
_inAudioIxRead += _frameLenSmpl * _channels;
return *bitStreamLenByte;
}
@ -159,6 +168,9 @@ int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
}
ret = WebRtcOpus_EncoderCreate(&_encoderInstPtr,
codecParams->codecInstant.channels);
// Store number of channels.
_channels = codecParams->codecInstant.channels;
if (ret < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Encoder creation failed for Opus");
@ -170,6 +182,10 @@ int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
"Setting initial bitrate failed for Opus");
return ret;
}
// Store bitrate.
_bitrate = codecParams->codecInstant.rate;
return 0;
}
@ -182,7 +198,14 @@ int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
codecParams->codecInstant.channels) < 0) {
return -1;
}
return WebRtcOpus_DecoderInit(_decoderInstPtr);
if (WebRtcOpus_DecoderInit(_decoderInstPtr) < 0) {
return -1;
}
if (WebRtcOpus_DecoderInitSlave(_decoderInstPtr) < 0) {
return -1;
}
return 0;
}
int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
@ -198,12 +221,26 @@ int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
// TODO(tlegrand): Decoder is registered in NetEQ as a 32 kHz decoder, which
// is true until we have a full 48 kHz system, and remove the downsampling
// in the Opus decoder wrapper.
SET_CODEC_PAR((codecDef), kDecoderOpus, codecInst.pltype, _decoderInstPtr,
32000);
SET_OPUS_FUNCTIONS((codecDef));
if (codecInst.channels == 1) {
SET_CODEC_PAR(codecDef, kDecoderOpus, codecInst.pltype, _decoderInstPtr,
32000);
} else {
SET_CODEC_PAR(codecDef, kDecoderOpus_2ch, codecInst.pltype,
_decoderInstPtr, 32000);
}
// If this is the master of NetEQ, regular decoder will be added, otherwise
// the slave decoder will be used.
if (_isMaster) {
SET_OPUS_FUNCTIONS(codecDef);
} else {
SET_OPUSSLAVE_FUNCTIONS(codecDef);
}
return 0;
}
ACMGenericCodec* ACMOpus::CreateInstance(void) {
return NULL;
}
@ -258,6 +295,23 @@ int16_t ACMOpus::SetBitRateSafe(const int32_t rate) {
return -1;
}
bool ACMOpus::IsTrueStereoCodec() {
return true;
}
// Copy the stereo packet so that NetEq will insert into both master and slave.
void ACMOpus::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
// Check for valid inputs.
assert(payload != NULL);
assert(*payload_length > 0);
// Duplicate the payload.
memcpy(&payload[*payload_length], &payload[0],
sizeof(uint8_t) * (*payload_length));
// Double the size of the packet.
*payload_length *= 2;
}
#endif // WEBRTC_CODEC_OPUS
} // namespace webrtc

5
webrtc/modules/audio_coding/main/source/acm_opus.h
View File

@ -50,10 +50,15 @@ class ACMOpus : public ACMGenericCodec {
int16_t SetBitRateSafe(const int32_t rate);
bool IsTrueStereoCodec();
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
WebRtcOpusEncInst* _encoderInstPtr;
WebRtcOpusDecInst* _decoderInstPtr;
uint16_t _sampleFreq;
uint16_t _bitrate;
int _channels;
};
} // namespace webrtc

3
webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
View File

@ -120,6 +120,9 @@
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'defines': [
'<@(audio_coding_defines)',
],
'sources': [
'../test/ACMTest.cc',
'../test/APITest.cc',