Adding Opus stereo support to WebRTC

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

webrtc/modules/audio_coding/main/source/acm_opus.cc

@@ -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