| /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 ||
|
| initial_delay_manager.h | 51 // since the last time |new_codec| should be true. |sample_rate_hz| is the
60 bool new_codec,
|
| 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,
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| codec_database.h | 158 VideoCodec* new_codec,
|
| codec_database.cc | 547 VideoCodec* new_codec,
550 assert(new_codec);
579 memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
|
| /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...] |
