/external/chromium_org/third_party/webrtc/ |
frame_callback.h | 24 EncodedFrame() : data_(NULL), length_(0), frame_type_(kFrameEmpty) {}
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common_types.h | 158 kFrameEmpty = 0,
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
video_receiver_unittest.cc | 94 header.frameType = kFrameEmpty; 118 header.frameType = kFrameEmpty; 133 header.frameType = kFrameEmpty; 169 header.frameType = kFrameEmpty; 194 header.frameType = kFrameEmpty;
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packet.cc | 27 frameType(kFrameEmpty), 87 frameType = kFrameEmpty;
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session_info.cc | 442 if (packet.frameType == kFrameEmpty) { 494 } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
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encoded_frame.cc | 173 return kFrameEmpty;
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frame_buffer.cc | 102 if (packet.frameType != kFrameEmpty) {
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receiver_unittest.cc | 69 (frame_type != kFrameEmpty) ? num_of_packets : 0, 70 (frame_type == kFrameEmpty) ? 1 : 0,
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decoding_state_unittest.cc | 184 packet.frameType = kFrameEmpty; 199 packet.frameType = kFrameEmpty;
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video_sender.cc | 359 if (_nextFrameTypes[0] == kFrameEmpty) {
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jitter_buffer_unittest.cc | 167 (frame_type != kFrameEmpty) ? 1 : 0, 168 (frame_type == kFrameEmpty) ? 1 : 0, 880 packet_->frameType = kFrameEmpty; [all...] |
session_info_unittest.cc | 182 packet_.frameType = kFrameEmpty; [all...] |
jitter_buffer.cc | 640 if (packet.frameType != kFrameEmpty) { [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_format_h264_unittest.cc | 92 kRtpVideoH264, max_payload_size, NULL, kFrameEmpty)); 166 RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty)); 195 RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty)); 232 RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty)); 267 RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty)); 315 RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
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rtp_sender_audio.cc | 277 if (frameType == kFrameEmpty) { 278 // kFrameEmpty is used to drive the DTMF when in CN mode 337 if (frameType == kFrameEmpty) {
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rtp_sender.cc | 32 case kFrameEmpty: return "empty"; 430 frame_type == kFrameEmpty); 439 if (frame_type == kFrameEmpty) [all...] |
rtp_sender_unittest.cc | [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
stream_generator.cc | 57 sequence_number_, timestamp_, 0, false, false, kFrameEmpty));
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/external/chromium_org/third_party/webrtc/video/ |
send_statistics_proxy.cc | 112 case kFrameEmpty:
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_receiver_unittest.cc | 49 last_frame_type_(kFrameEmpty) { 125 if (frame_type == kFrameEmpty)
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acm_receiver_unittest_oldapi.cc | 49 last_frame_type_(kFrameEmpty) { 129 if (frame_type == kFrameEmpty)
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/external/webrtc/src/ |
common_types.h | 215 kFrameEmpty = 0,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
Channel.cc | 42 if (frameType == kFrameEmpty) {
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TestAllCodecs.cc | 72 if (frame_type == kFrameEmpty) {
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TestStereo.cc | 61 if (frame_type == kFrameEmpty) {
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