Searched refs:new_codec (Results 1 - 5 of 5) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/acm2/
H A Dinitial_delay_manager.cc38 bool new_codec,
43 // If payload of audio packets is changing |new_codec| has to be true.
44 assert(!(!new_codec && type == kAudioPacket &&
64 if (new_codec ||
34 UpdateLastReceivedPacket( const WebRtcRTPHeader& rtp_info, uint32_t receive_timestamp, PacketType type, bool new_codec, int sample_rate_hz, SyncStream* sync_stream) argument
H A Dinitial_delay_manager.h51 // since the last time |new_codec| should be true. |sample_rate_hz| is the
60 bool new_codec,
/external/webrtc/webrtc/modules/video_coding/
H A Dcodec_database.h132 VideoCodec* new_codec) const;
H A Dcodec_database.cc531 VideoCodec* new_codec) const {
533 assert(new_codec);
568 memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
/external/webrtc/talk/app/webrtc/
H A Dwebrtcsdp.cc2382 U new_codec = GetCodecWithPayloadType(static_cast<T*>(content_desc)->codecs(), local
2384 AddParameters(parameters, &new_codec);
2385 AddOrReplaceCodec<T, U>(content_desc, new_codec);
2394 U new_codec = GetCodecWithPayloadType(static_cast<T*>(content_desc)->codecs(), local
2396 AddFeedbackParameter(feedback_param, &new_codec);
2397 AddOrReplaceCodec<T, U>(content_desc, new_codec);

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