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/frameworks/av/media/libstagefright/codecs/amrwbenc/inc/
main.h	30 Word16 speech16k[], /* input : 320 new speech samples (at 16 kHz) */
wb_vad_c.h	35 /* constants for speech level estimation */ 42 #define SPEECH_LEVEL_INIT NOM_LEVEL /* initial speech level */
/frameworks/av/media/libstagefright/codecs/amrwbenc/src/
wb_vad.c	343 * Purpose : Add hangover after speech bursts 506 * subtracting MIN_SPEECH_SNR*noise_level from speech level */ 559 * Purpose : Estimate speech level 562 * The speech frames must locate within SP_EST_COUNT number of frames. 596 /* update speech estimate */ 597 tmp = (st->sp_max >> 1); /* scale to get "average" speech level */ 611 /* clear all counters used for speech estimation */ 759 Word16 wb_vad( /* Return value : VAD Decision, 1 = speech, 0 = noise */ 802 Estimate_Speech(st, temp); /* Estimate speech level */
bits.c	143 /* sort and pack AMR-WB speech or SID bits */ 161 /* insert SID type indication and speech mode in case of SID frame */
/frameworks/base/docs/html/sdk/api_diff/23/changes/
packages_index_changes.html	89 <A HREF="pkg_android.speech.html" class="hiddenlink" target="rightframe">android.speech</A><br> 90 <A HREF="pkg_android.speech.tts.html" class="hiddenlink" target="rightframe">android.speech.tts</A><br>
/frameworks/base/packages/Keyguard/src/com/android/keyguard/
ObscureSpeechDelegate.java	32 * Accessibility delegate that obscures speech for a view when the user has
/frameworks/ex/common/java/com/android/common/userhappiness/
UserHappinessSignals.java	21 import com.android.common.speech.LoggingEvents;
/frameworks/support/v17/leanback/src/android/support/v17/leanback/app/
SearchFragment.java	21 import android.speech.SpeechRecognizer; 22 import android.speech.RecognizerIntent; 51 * {@link #setSpeechRecognitionCallback(SpeechRecognitionCallback)}, an internal speech 56 * Speech recognition is automatically started when fragment is created, but 380 * Starts speech recognition. Typical use case is that 382 * Note that SearchFragment automatically starts speech recognition 481 * Sets this callback to have the fragment pass speech recognition requests 521 * @param intent Intent received from a speech recognition service. 532 * Returns an intent that can be used to request speech recognition.
SearchSupportFragment.java	23 import android.speech.SpeechRecognizer; 24 import android.speech.RecognizerIntent; 53 * {@link #setSpeechRecognitionCallback(SpeechRecognitionCallback)}, an internal speech 58 * Speech recognition is automatically started when fragment is created, but 382 * Starts speech recognition. Typical use case is that 384 * Note that SearchSupportFragment automatically starts speech recognition 483 * Sets this callback to have the fragment pass speech recognition requests 523 * @param intent Intent received from a speech recognition service. 534 * Returns an intent that can be used to request speech recognition.
/packages/apps/CellBroadcastReceiver/src/com/android/cellbroadcastreceiver/
CellBroadcastAlertDialog.java	28 * Alert audio and text-to-speech handled by {@link CellBroadcastAlertAudio}.
/packages/apps/QuickSearchBox/src/com/android/quicksearchbox/
VoiceSearch.java	27 import android.speech.RecognizerIntent;
SearchWidgetProvider.java	20 import com.android.common.speech.Recognition; 36 import android.speech.RecognizerIntent;
/frameworks/av/media/libstagefright/codecs/amrnb/enc/src/
dtx_enc.cpp	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 55 the calculation of the sum of squares of speech signals in 188 dtx_enc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 301 dtx_enc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 427 dtx_enc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 527 dtx_enc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 839 speech = vector of speech samples of type Word16; vector length is 868 dtx_enc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 875 Word16 speech[] // i : speech sample     [all...]
ets_to_wmf.cpp	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 93 frame_type_3gpp = decoder speech bit rate (enum Frame_Type_3GPP) 94 ets_input_ptr = pointer to input encoded speech bits in ETS format (Word16) 95 wmf_output_ptr = pointer to output encoded speech bits in WMF format(UWord8) 98 wmf_output_ptr = pointer to encoded speech bits in the WMF format (UWord8) 108 encoded speech bits when converting from ETS to WMF or IF2 120 multimedia forum). ETS format has the encoded speech bits each separate with 124 encoded speech bits. The final byte has padded zeros to make the frame byte
spstproc.cpp	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 101 speech -- Pointer to Word16 -- speech segment 143 spstproc.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 173 Word16 *speech, /* i : speech segment */ 233 * - find synthesis speech corresponding to exc[] * 284 mem_err[j] = speech[i_subfr + i] - synth[i_subfr + i];
dtx_enc.h	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 181 Word16 speech[], /* i : speech samples */ 189 * Purpose : adds extra speech hangover to analyze speech on the decoding side.
/frameworks/av/media/libstagefright/codecs/amrwb/src/
dtx_decoder_amr_wb.cpp	22 ANSI-C code for the Adaptive Multi-Rate - Wideband (AMR-WB) speech codec 156 st->dtxGlobalState = SPEECH; 171 frame_type | SPEECH | DTX | DTX_MUTE 174 RX_SPEECH_PR_DEGRADED | SPEECH | SPEECH | SPEECH 176 RX_SPEECH_BAD, | SPEECH | DTX | DTX_MUTE 184 RX_NO_DATA, | SPEECH | DTX/(DTX_MUTE)| DTX_MUTE 215 /* This function is called if synthesis state is not SPEECH the globally passed inputs to this function 216 * are st->sid_frame st->valid_data st->dtxHangoverAdded new_state (SPEECH, DTX, DTX_MUTE) *     [all...]
/frameworks/av/media/libstagefright/codecs/amrnb/dec/src/
post_pro.cpp	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 91 speech. Post-processing include filtering the output speech through a second 93 output speech by a factor of 2. In addition to the post-processing function 169 post_pro.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 264 This function performs post-processing on the output speech signal. First, 265 the output speech goes through a second order high pass filter with a 266 cutoff frequency of 60 Hz. Then, the filtered output speech is multiplied 280 post_pro.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 310 //Multiplication by two of output speech with saturation     [all...]
dec_amr.cpp	22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec 40 This file contains the function used to decode one speech frame using a given 151 dec_amr.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 304 dec_amr.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 498 synth = buffer containing synthetic speech (Word16) 504 synth buffer contains the decoded speech samples 519 This function performs the decoding of one speech frame for a given codec 530 dec_amr.c, UMTS GSM AMR speech codec, R99 - Version 3.2.0, March 2, 2001 541 Word16 synth[], // o : synthesis speech (L_FRAME) 588 enum DTXStateType newDTXState; // SPEECH , DTX, DTX_MUT     [all...]
/external/libgsm/man/
gsm_option.3	22 standard for full-rate speech transcoding, a lossy 23 speech compression algorithm.
/external/libopus/
README	3 Opus is a codec for interactive speech and audio transmission over the Internet. 7 performances. It can scale from low bit-rate narrowband speech to very high
/external/svox/pico/lib/
picokfst.h	67 PICOKFST_PLANE_POS = 5, /* part of speech plane */ 77 PICOKFST_TRANSMODE_POSUSED = 2 /* FST contains Part Of Speech symbols */
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/
lattice_neon.S	29 @ instructions, smulwb, and smull. Speech quality was not degraded by 30 @ testing speech and tone vectors.
/external/webrtc/src/
common_types.h	260 // fraction (of original stream) of synthesized speech inserted through 263 // fraction of synthesized speech inserted through pre-emptive expansion 287 StatVal speech_rx; // long-term speech levels on receiving side 288 StatVal speech_tx; // long-term speech levels on transmitting side 401 // BGN is not used at all. Silence is produced after speech extrapolation
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/
test_iSACfixfloat.c	178 printf("infile : Normal speech input file\n\n"); 179 printf("outfile : Speech output file\n\n"); 433 /* Read 10 ms speech block */ 643 /* Write decoded speech frame to file */ 674 printf("\n\nLength of speech file: %.1f s\n", length_file);

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