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