| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| time_stretch_unittest.cc | 66 audio_(new int16_t[block_size_]),
72 RTC_CHECK(input_file_->Read(block_size_, audio_.get()));
73 return audio_.get();
106 rtc::scoped_ptr<int16_t[]> audio_; member in class:webrtc::TimeStretchTest
|
| /external/webrtc/talk/media/base/ |
| streamparams.cc | 57 return GetStream(audio_, selector, stream);
71 audio_ = streams.audio_;
77 AddStream(&audio_, stream);
90 return RemoveStream(&audio_, selector);
|
| streamparams.h | 216 return audio_.empty() && video_.empty() && data_.empty();
219 std::vector<StreamParams>* mutable_audio() { return &audio_; }
222 const std::vector<StreamParams>& audio() const { return audio_; }
243 std::vector<StreamParams> audio_; member in struct:cricket::MediaStreams
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red_unittest.cc | 44 memset(audio_, 0, sizeof(audio_));
65 rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
73 int16_t audio_[kMaxNumSamples]; member in class:webrtc::AudioEncoderCopyRedTest
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng_unittest.cc | 41 memset(audio_, 0, kMaxNumSamples * 2);
80 rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
192 int16_t audio_[kMaxNumSamples]; member in class:webrtc::AudioEncoderCngTest
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender.cc | 135 audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
333 ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
370 return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
459 if (audio_->RED(&red_pl_type) == 0) {
532 ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
[all...] |
| rtcp_sender.cc | 146 : audio_(audio),
218 (audio_ ? RTCP_INTERVAL_AUDIO_MS / 2 : RTCP_INTERVAL_VIDEO_MS / 2);
414 if (!audio_ && sendKeyframeBeforeRTP) {
883 if (!audio_) {
[all...] |
| rtcp_sender.h | 201 const bool audio_; member in class:webrtc::RTCPSender
|
| rtp_rtcp_impl.h | 358 const bool audio_; member in class:webrtc::ModuleRtpRtcpImpl
|
| rtp_sender.h | 395 rtc::scoped_ptr<RTPSenderAudio> audio_; member in class:webrtc::RTPSender
|
| rtp_rtcp_impl.cc | 85 audio_(configuration.audio),
965 if (audio_)
|
| /prebuilts/misc/common/frameworks/ |
| platformprotos-prebuilt.jar | |
