| 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
|
|
|