/third_party/ltp/testcases/kernel/device-drivers/v4l/user_space/ |
H A D | test_VIDIOC_ENUMAUDIO.c | 41 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO() local 47 memset(&audio, 0xff, sizeof(audio)); in test_VIDIOC_ENUMAUDIO() 48 audio.index = i; in test_VIDIOC_ENUMAUDIO() 49 ret_enum = ioctl(get_video_fd(), VIDIOC_ENUMAUDIO, &audio); in test_VIDIOC_ENUMAUDIO() 58 CU_ASSERT_EQUAL(audio.index, i); in test_VIDIOC_ENUMAUDIO() 60 CU_ASSERT(0 < strlen((char *)audio.name)); in test_VIDIOC_ENUMAUDIO() 62 ((char *)audio.name, sizeof(audio.name))); in test_VIDIOC_ENUMAUDIO() 64 //CU_ASSERT_EQUAL(audio in test_VIDIOC_ENUMAUDIO() 110 struct v4l2_audio audio; test_VIDIOC_ENUMAUDIO_S32_MAX() local 130 struct v4l2_audio audio; test_VIDIOC_ENUMAUDIO_S32_MAX_1() local 150 struct v4l2_audio audio; test_VIDIOC_ENUMAUDIO_U32_MAX() local 171 struct v4l2_audio audio; test_VIDIOC_ENUMAUDIO_NULL() local [all...] |
H A D | test_VIDIOC_AUDIO.c | 67 struct v4l2_audio audio; in test_VIDIOC_G_AUDIO() local 70 memset(&audio, 0xff, sizeof(audio)); in test_VIDIOC_G_AUDIO() 71 ret_get = ioctl(get_video_fd(), VIDIOC_G_AUDIO, &audio); in test_VIDIOC_G_AUDIO() 80 //CU_ASSERT_EQUAL(audio.index, ?); in test_VIDIOC_G_AUDIO() 82 CU_ASSERT(0 < strlen((char *)audio.name)); in test_VIDIOC_G_AUDIO() 83 CU_ASSERT(valid_string((char *)audio.name, sizeof(audio.name))); in test_VIDIOC_G_AUDIO() 85 CU_ASSERT(valid_audio_capability(audio.capability)); in test_VIDIOC_G_AUDIO() 86 CU_ASSERT(valid_audio_mode(audio in test_VIDIOC_G_AUDIO() 129 struct v4l2_audio audio; test_VIDIOC_G_AUDIO_ignore_index() local 166 struct v4l2_audio audio; test_VIDIOC_G_AUDIO_NULL() local 296 struct v4l2_audio audio; test_VIDIOC_S_AUDIO_S32_MAX() local 353 struct v4l2_audio audio; test_VIDIOC_S_AUDIO_S32_MAX_1() local 410 struct v4l2_audio audio; test_VIDIOC_S_AUDIO_U32_MAX() local [all...] |
/third_party/ffmpeg/tests/fate/ |
H A D | lossless-audio.mak | 2 fate-lossless-alac: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/inside.m4a -f s16le -af aresample 5 fate-lossless-meridianaudio: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.mlp -f s16le 8 fate-ralf: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.rmvb -vn -f s16le -af aresample 11 fate-lossless-shorten: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.shn -f s16le -af aresample 14 fate-lossless-tak: CMD = crc -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.tak -af aresample 17 fate-lossless-tta: CMD = crc -i $(TARGET_SAMPLES)/lossless-audio/inside.tta 20 fate-lossless-tta-encrypted: CMD = crc -password ffmpeg -i $(TARGET_SAMPLES)/lossless-audio/encrypted.tta 23 fate-lossless-wma: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.wma -f s16le -frames 209 -af aresample 24 fate-lossless-wma24-1: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/master_audio_2.0_24bit.wma -f s24le -af aresample 25 fate-lossless-wma24-2: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/Mega_Weird_Audio_Test_24bi [all...] |
H A D | audio.mak | 16 FATE_SAMPLES_AUDIO-$(call DEMDEC, BMV, BMV_AUDIO) += fate-bmv-audio 17 fate-bmv-audio: CMD = framecrc -i $(TARGET_SAMPLES)/bmv/SURFING-partial.BMV -vn 19 FATE_SAMPLES_AUDIO-$(call DEMDEC, DSICIN, DSICINAUDIO) += fate-delphine-cin-audio 20 fate-delphine-cin-audio: CMD = framecrc -i $(TARGET_SAMPLES)/delphine-cin/LOGO-partial.CIN -vn 58 FATE_SAMPLES_AUDIO-$(call DEMDEC, PAF, PAF_AUDIO) += fate-paf-audio 59 fate-paf-audio: CMD = framecrc -i $(TARGET_SAMPLES)/paf/hod1-partial.paf -vn 61 FATE_SAMPLES_AUDIO-$(call DEMDEC, VMD, VMDAUDIO) += fate-sierra-vmd-audio 62 fate-sierra-vmd-audio: CMD = framecrc -i $(TARGET_SAMPLES)/vmd/12.vmd -vn -af aresample 64 FATE_SAMPLES_AUDIO-$(call DEMDEC, SMACKER, SMACKAUD) += fate-smacker-audio 65 fate-smacker-audio [all...] |
H A D | aac.mak | 162 fate-aac-ln-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_coder fast -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 512k -fflags +bitexact -flags +bitexact 164 fate-aac-ln-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 171 fate-aac-ln-encode-128k: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_coder fast -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact 173 fate-aac-ln-encode-128k: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 180 fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_coder fast -aac_pns 1 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact 182 fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 189 fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_coder fast -aac_tns 1 -aac_is 0 -aac_pns 0 -aac_ms 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact 191 fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 198 fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_coder fast -aac_pns 0 -aac_is 1 -aac_ms 0 -b:a 128k -aac_tns 0 -cutoff 22050 -fflags +bitexact -flags +bitexact 200 fate-aac-is-encode: REF = $(SAMPLES)/audio [all...] |
H A D | monkeysaudio.mak | 5 fate-lossless-monkeysaudio-$(1)-normal: CMD = crc -auto_conversion_filters -i $(TARGET_SAMPLES)/lossless-audio/luckynight-mac$(1)-c2000.ape -af atrim=end_sample=73728 9 fate-lossless-monkeysaudio-$(1)-extrahigh: CMD = crc -auto_conversion_filters -i $(TARGET_SAMPLES)/lossless-audio/luckynight-mac$(1)-c4000.ape -af atrim=end_sample=73728 17 fate-lossless-monkeysaudio-399: CMD = md5 -i $(TARGET_SAMPLES)/lossless-audio/luckynight-partial.ape -f s16le -af aresample
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H A D | flac.mak | 15 fate-flac-16-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 20 fate-flac-24-%: REF = $(SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav 23 fate-flac-rice-params: REF = $(SAMPLES)/audio-reference/chorusnoise_2ch_44kHz_s16.wav
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H A D | als.mak | 5 fate-mpeg4-als-conformance-$(1): CMD = crc -i $(TARGET_SAMPLES)/lossless-audio/als_$(1)_2ch48k16b.mp4 12 fate-mpeg4-als-conformance-09: CMD = crc -i $(TARGET_SAMPLES)/lossless-audio/als_09_512ch2k16b.mp4
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H A D | caf.mak | 5 fate-caf-alac-remux: CMD = transcode m4a $(TARGET_SAMPLES)/lossless-audio/inside.m4a caf "-map 0:a -c copy -metadata major_brand= " "-c copy -t 0.2" "-show_entries format_tags" 14 fate-caf-pcm_s24le-remux: CMD = transcode wav $(TARGET_SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav caf "-c copy" "-c copy -t 0.05" 19 fate-caf-pcm_s24-remux: CMD = transcode wav $(TARGET_SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav caf "-c pcm_s24be" "-c copy -t 0.05"
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H A D | alac.mak | 14 fate-alac-16-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav 17 fate-alac-24-%: REF = $(SAMPLES)/audio-reference/divertimenti_2ch_96kHz_s24.wav
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H A D | truehd.mak | 2 fate-truehd-5.1: CMD = md5pipe -f truehd -i $(TARGET_SAMPLES)/lossless-audio/truehd_5.1.raw -f s32le 7 fate-truehd-5.1-downmix-2.0: CMD = md5pipe -f truehd -request_channel_layout FL+FR -i $(TARGET_SAMPLES)/lossless-audio/truehd_5.1.raw -f s32le
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H A D | webm-dash-manifest.mak | 7 FATE_WEBM_DASH_MANIFEST += fate-webm-dash-manifest-unaligned-audio-streams 8 fate-webm-dash-manifest-unaligned-audio-streams: CMD = run $(FFMPEG) -nostdin -f webm_dash_manifest -i $(TARGET_SAMPLES)/vp8/dash_audio1.webm -f webm_dash_manifest -i $(TARGET_SAMPLES)/vp8/dash_audio3.webm -c copy -map 0 -map 1 -f webm_dash_manifest -adaptation_sets "id=0,streams=0,1" -
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/third_party/ffmpeg/libavformat/ |
H A D | xmv.c | 72 /** An audio packet with an XMV file. */ 75 int stream_index; ///< The decoder stream index for this audio packet. 89 uint32_t data_size; ///< The size of the remaining audio data. 90 uint64_t data_offset; ///< The offset of the audio data within the file. 92 uint32_t frame_size; ///< Number of bytes to put into an audio frame. 94 uint64_t block_count; ///< Running counter of decompressed audio block. 99 uint16_t audio_track_count; ///< Number of audio track in this file. 115 XMVAudioPacket *audio; ///< The audio packets contained in each packet. member 139 av_freep(&xmv->audio); in xmv_read_close() 441 XMVAudioPacket *audio = &xmv->audio[stream]; xmv_fetch_audio_packet() local [all...] |
H A D | c93.c | 43 AVStream *audio; member 79 /* Audio streams are added if audio packets are found */ in read_header() 119 if (!c93->audio) { in read_packet() 120 c93->audio = avformat_new_stream(s, NULL); in read_packet() 121 if (!c93->audio) in read_packet() 123 c93->audio->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; in read_packet() 126 ret = ff_voc_get_packet(s, pkt, c93->audio, datasize - 26); in read_packet()
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H A D | matroskadec.c | 226 /* real audio header (extracted from extradata) */ 272 MatroskaTrackAudio audio; member 611 { MATROSKA_ID_TRACKAUDIO, EBML_NEST, 0, 0, offsetof(MatroskaTrack, audio), { .n = matroska_track_audio } }, 2476 if (track->audio.samplerate < 0 || track->audio.samplerate > INT_MAX || in matroska_parse_tracks() 2477 isnan(track->audio.samplerate)) { in matroska_parse_tracks() 2480 track->audio.samplerate); in matroska_parse_tracks() 2481 track->audio.samplerate = 8000; in matroska_parse_tracks() 2502 if (!track->audio.out_samplerate) in matroska_parse_tracks() 2503 track->audio in matroska_parse_tracks() [all...] |
H A D | segafilmenc.c | 86 /* Always the same, carries no more information than "this is audio" */ in film_write_packet() 128 av_log(format_context, AV_LOG_ERROR, "Sega FILM allows a maximum of one audio stream.\n"); in film_init() 133 "Incompatible audio stream format.\n"); in film_init() 236 AVStream *audio = format_context->streams[film->audio_index]; in film_write_header() local 237 int audio_codec = get_audio_codec_id(audio->codecpar->codec_id); in film_write_header() 239 bytestream_put_byte(&ptr, audio->codecpar->ch_layout.nb_channels); /* Audio channels */ in film_write_header() 240 bytestream_put_byte(&ptr, audio->codecpar->bits_per_coded_sample); /* Audio bit depth */ in film_write_header() 242 bytestream_put_be16(&ptr, audio->codecpar->sample_rate); /* Audio sampling rate */ in film_write_header() 244 /* If there is no audio, all the audio field in film_write_header() [all...] |
H A D | mpeg.c | 68 int audio = 0, invalid = 0, score = 0; in mpegps_probe() local 87 // and audio streams in mpegps_probe() 88 else if ((code & 0xe0) == AUDIO_ID && pes) {audio++; i+=len;} in mpegps_probe() 98 if (vid + audio > invalid + 1) /* invalid VDR files nd short PES streams */ in mpegps_probe() 102 // vid, audio, sys, pspack, invalid, p->buf_size); in mpegps_probe() 105 return (audio > 12 || vid > 3 || pspack > 2) ? AVPROBE_SCORE_EXTENSION + 2 in mpegps_probe() 106 : AVPROBE_SCORE_EXTENSION / 2 + (audio + vid + pspack > 1); // 1 more than mp3 in mpegps_probe() 107 if (pspack > invalid && (priv1 + vid + audio) * 10 >= pspack * 9) in mpegps_probe() 110 if ((!!vid ^ !!audio) && (audio > in mpegps_probe() [all...] |
/third_party/libsnd/tests/ |
H A D | chunk_test.c | 325 short audio [16] ; in wav_subchunk_test() local 366 /* Add some audio data. */ in wav_subchunk_test() 367 memset (audio, 0, sizeof (audio)) ; in wav_subchunk_test() 368 sf_write_short (file, audio, ARRAY_LEN (audio)) ; in wav_subchunk_test() 375 sfinfo.frames != ARRAY_LEN (audio), in wav_subchunk_test() 376 "\n\nLine %d : Incorrect sample count (%d should be %d)\n", __LINE__, (int) sfinfo.frames, (int) ARRAY_LEN (audio) in wav_subchunk_test() 394 short audio [16] ; in large_free_test() local 426 /* Add some audio dat in large_free_test() [all...] |
/third_party/ffmpeg/libavcodec/ |
H A D | libopusenc.c | 462 uint8_t *audio; in libopus_encode() local 471 audio = opus->samples; in libopus_encode() 473 audio, frame->data[0], opus->encoder_channel_map, in libopus_encode() 476 audio = opus->samples; in libopus_encode() 477 memcpy(audio, frame->data[0], frame->nb_samples * sample_size); in libopus_encode() 479 audio = frame->data[0]; in libopus_encode() 483 audio = opus->samples; in libopus_encode() 484 memset(audio, 0, opus->opts.packet_size * sample_size); in libopus_encode() 494 ret = opus_multistream_encode_float(opus->enc, (float *)audio, in libopus_encode() 498 ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio, in libopus_encode() [all...] |
H A D | g723_1dec.c | 446 int16_t *buf = p->audio + LPC_ORDER + offset; in comp_ppf_coeff() 520 int16_t *buf = p->audio + LPC_ORDER; in comp_interp_index() 845 vector_ptr = p->audio + LPC_ORDER; in generate_noise() 924 memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN, in generate_noise() 959 int16_t *audio = p->audio; in g723_1_decode_frame() local 1021 ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i, in g723_1_decode_frame() 1028 audio = vector_ptr - LPC_ORDER; in g723_1_decode_frame() 1045 int16_t *buf = p->audio + LPC_ORDER; in g723_1_decode_frame() 1077 memcpy(p->audio, in g723_1_decode_frame() [all...] |
H A D | psymodel.c | 2 * audio encoder psychoacoustic model 139 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels) in ff_psy_preprocess() argument 148 &audio[ch][frame_size], 1, &audio[ch][frame_size], 1); in ff_psy_preprocess()
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H A D | dca_core.c | 104 av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", h.audio_mode); in parse_frame_header() 108 av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n"); in parse_frame_header() 152 // 5.3.2 - Primary audio coding header 166 // Number of primary audio channels in parse_coding_header() 169 av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode); in parse_coding_header() 400 // 5.4.1 - Primary audio coding side information 528 static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio) in decode_blockcodes() argument 535 audio[n] = code1 - div * levels - offset; in decode_blockcodes() 540 audio[ in decode_blockcodes() 548 parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits) parse_block_codes() argument 564 parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel) parse_huffman_codes() argument 575 extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch) extract_audio() argument 626 int32_t audio[16], scale; parse_subframe_audio() local 1023 int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale; parse_xbr_subframe() local 1213 int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale; parse_x96_subframe_audio() local [all...] |
H A D | psymodel.h | 2 * audio encoder psychoacoustic model 122 * @param audio samples for the current frame 129 FFPsyWindowInfo (*window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type); 181 * This should be moved into some audio filter eventually. * 186 * psychoacoustic model audio preprocessing initialization 191 * Preprocess several channel in audio frame in order to compress it better. 194 * @param audio samples to be filtered (in place) 197 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels); 200 * Cleanup audio preprocessing module.
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H A D | aacenc.c | 92 * Make AAC audio config object. 139 const float *audio) 147 fdsp->vector_fmul (out, audio, lwindow, 1024); in WINDOW_FUNC() 148 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024); in WINDOW_FUNC() 157 fdsp->vector_fmul(out, audio, lwindow, 1024); in WINDOW_FUNC() 158 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448); in WINDOW_FUNC() 159 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128); in WINDOW_FUNC() 170 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128); in WINDOW_FUNC() 171 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); in WINDOW_FUNC() 172 fdsp->vector_fmul_reverse(out + 1024, audio in WINDOW_FUNC() 201 apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio) apply_window_and_mdct() argument [all...] |
/third_party/ffmpeg/libavfilter/ |
H A D | audio.c | 27 #include "audio.h" 104 if (link->dstpad->get_buffer.audio) in ff_get_audio_buffer() 105 ret = link->dstpad->get_buffer.audio(link, nb_samples); in ff_get_audio_buffer()
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