/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
H A D | audio_decoder_pcm.cc | 24 size_t encoded_len, 30 size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type); 36 size_t encoded_len) const { 38 return static_cast<int>(encoded_len / Channels()); 48 size_t encoded_len, 54 size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type); 60 size_t encoded_len) const { 62 return static_cast<int>(encoded_len / Channels()); 23 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument 47 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument
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/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
H A D | audio_decoder_pcm16b.cc | 30 size_t encoded_len, 37 size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded); 43 size_t encoded_len) const { 45 return static_cast<int>(encoded_len / (2 * Channels())); 29 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | constant_pcm_packet_source.cc | 34 size_t encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_); local 35 RTC_CHECK_EQ(2U, encoded_len);
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/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | audio_decoder_ilbc.cc | 32 size_t encoded_len, 38 int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded, 31 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
H A D | audio_decoder.cc | 20 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, argument 24 int duration = PacketDuration(encoded, encoded_len); 29 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, 33 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, argument 37 int duration = PacketDurationRedundant(encoded, encoded_len); 42 return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, 47 size_t encoded_len, 50 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, 71 size_t encoded_len) const { 76 size_t encoded_len) cons 46 DecodeRedundantInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
H A D | audio_decoder_g722.cc | 34 size_t encoded_len, 41 WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type); 51 size_t encoded_len) const { 53 return static_cast<int>(2 * encoded_len / Channels()); 73 size_t encoded_len, 80 uint8_t* encoded_deinterleaved = new uint8_t[encoded_len]; 81 SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved); 84 encoded_len / 2, decoded, &temp_type); 86 dec_state_right_, &encoded_deinterleaved[encoded_len / 2], 87 encoded_len / 33 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument 72 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument 114 SplitStereoPacket(const uint8_t* encoded, size_t encoded_len, uint8_t* encoded_deinterleaved) argument [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
H A D | audio_decoder_isac_t_impl.h | 43 size_t encoded_len, 55 T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type); 42 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
H A D | audio_decoder_opus.cc | 29 size_t encoded_len, 36 WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type); 44 size_t encoded_len, 48 if (!PacketHasFec(encoded, encoded_len)) { 50 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, 56 int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded, 69 size_t encoded_len) const { 70 return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len); 74 size_t encoded_len) const { 75 if (!PacketHasFec(encoded, encoded_len)) { 28 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument 43 DecodeRedundantInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | audio_decoder_impl.cc | 69 size_t encoded_len, 68 DecodeInternal(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, SpeechType* speech_type) argument
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/external/boringssl/src/crypto/base64/ |
H A D | base64_test.cc | 318 const size_t encoded_len = strlen(t->encoded); local 320 std::vector<uint8_t> out(encoded_len); 322 for (size_t chunk_size = 1; chunk_size <= encoded_len; chunk_size++) { 327 for (size_t i = 0; i < encoded_len;) { 328 size_t todo = encoded_len - i; 345 if (i == encoded_len || 346 (i + 1 == encoded_len && t->encoded[i] == '\n') ||
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/external/webrtc/webrtc/base/ |
H A D | base64_unittest.cc | 919 size_t consumed = 0, encoded_len = strlen(encoded); local 920 bool success = Base64::DecodeFromArray(encoded, encoded_len, flags, 922 size_t unparsed = encoded_len - consumed;
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/external/wpa_supplicant_8/wpa_supplicant/ |
H A D | config_file.c | 291 size_t encoded_len = 0, len; local 303 nencoded = os_realloc(encoded, encoded_len + len); 311 os_memcpy(encoded + encoded_len, pos, len); 312 encoded_len += len; 328 blob->data = base64_decode(encoded, encoded_len, &blob->len);
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/external/wpa_supplicant_8/hostapd/src/wps/ |
H A D | wps_er.c | 902 size_t encoded_len; local 907 &encoded_len); 912 encoded_len = 0; 915 buf = wpabuf_alloc(1000 + encoded_len);
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/external/wpa_supplicant_8/src/wps/ |
H A D | wps_er.c | 902 size_t encoded_len; local 907 &encoded_len); 912 encoded_len = 0; 915 buf = wpabuf_alloc(1000 + encoded_len);
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/external/wpa_supplicant_8/wpa_supplicant/src/wps/ |
H A D | wps_er.c | 902 size_t encoded_len; local 907 &encoded_len); 912 encoded_len = 0; 915 buf = wpabuf_alloc(1000 + encoded_len);
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/external/boringssl/src/ssl/ |
H A D | ssl_test.cc | 692 size_t encoded_len; local 695 if (!SSL_SESSION_to_bytes(session.get(), &encoded_raw, &encoded_len)) { 700 if (encoded_len != input.size() || 704 hexdump(stderr, "After: ", encoded_raw, encoded_len);
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_unittest_oldapi.cc | 985 size_t encoded_len, 989 return decoder_->Decode(encoded, encoded_len, sample_rate_hz, 990 decoder_->PacketDuration(encoded, encoded_len) * 984 Decode(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, AudioDecoder::SpeechType* speech_type) argument
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/external/openssh/ |
H A D | sshkey.c | 3150 size_t encoded_len; local 3174 encoded_len = sshbuf_len(blob); 3175 if (encoded_len < (MARK_BEGIN_LEN + MARK_END_LEN) || 3181 encoded_len -= MARK_BEGIN_LEN; 3184 while (encoded_len > 0) { 3190 encoded_len--; 3193 if (encoded_len >= MARK_END_LEN && 3202 if (encoded_len == 0) {
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