/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
initial_delay_manager.cc | 38 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 ||
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initial_delay_manager.h | 51 // since the last time |new_codec| should be true. |sample_rate_hz| is the 60 bool new_codec,
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acm_receiver.cc | 269 bool new_codec = false; local 301 new_codec = true; 329 rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
codec_database.h | 158 VideoCodec* new_codec,
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codec_database.cc | 547 VideoCodec* new_codec, 550 assert(new_codec); 579 memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvideoengine.cc | 2932 webrtc::VideoCodec new_codec = *send_codec_; local 2947 webrtc::VideoCodec new_codec = *send_codec_; local 3014 webrtc::VideoCodec new_codec = *send_codec_; local [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
webrtcsdp.cc | 2346 U new_codec = GetCodec(static_cast<T*>(content_desc)->codecs(), payload_type); local 2357 U new_codec = GetCodec(static_cast<T*>(content_desc)->codecs(), payload_type); local [all...] |