| Index: webrtc/modules/audio_coding/neteq/tools/packet_source.cc | 
| diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.cc b/webrtc/modules/audio_coding/neteq/tools/packet_source.cc | 
| similarity index 57% | 
| copy from webrtc/modules/audio_coding/neteq/tools/audio_sink.cc | 
| copy to webrtc/modules/audio_coding/neteq/tools/packet_source.cc | 
| index fb848bdbd9b53e62af0c2a8cfb7fb3668a8e75c5..d6cb37ef39cc9ea4af170d7d8520602ae515be6e 100644 | 
| --- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.cc | 
| +++ b/webrtc/modules/audio_coding/neteq/tools/packet_source.cc | 
| @@ -8,14 +8,23 @@ | 
| *  be found in the AUTHORS file in the root of the source tree. | 
| */ | 
|  | 
| -#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" | 
| +#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" | 
|  | 
| namespace webrtc { | 
| namespace test { | 
|  | 
| -bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) { | 
| -  return left_sink_->WriteArray(audio, num_samples) && | 
| -         right_sink_->WriteArray(audio, num_samples); | 
| +PacketSource::PacketSource() : use_ssrc_filter_(false), ssrc_(0) {} | 
| + | 
| +PacketSource::~PacketSource() = default; | 
| + | 
| +void PacketSource::FilterOutPayloadType(uint8_t payload_type) { | 
| +  filter_.set(payload_type, true); | 
| } | 
| + | 
| +void PacketSource::SelectSsrc(uint32_t ssrc) { | 
| +  use_ssrc_filter_ = true; | 
| +  ssrc_ = ssrc; | 
| +} | 
| + | 
| }  // namespace test | 
| }  // namespace webrtc | 
|  |