search for: sample_r

...ig 2003-05-20 21:57:04.000000000 +0200 > +++ plugin.c 2003-07-08 22:03:37.000000000 +0200 > @@ -537,7 +537,7 @@ > file_info->bits_per_sample = metadata->data.stream_info.bits_per_sample; > file_info->channels = metadata->data.stream_info.channels; > file_info->sample_rate = metadata->data.stream_info.sample_rate; > - file_info->length_in_msec = file_info->total_samples * 10 / (file_info->sample_rate / 100); > + file_info->length_in_msec = (FLAC__uint64)file_info->total_samples * 10 / (file_info->sample_rate / 100); > } > els...

...requencies result in longer waves. But that?s not all: as frequencies increase, it appears that wavelengths increase to infinite length, then get shorter again as the wave reverses, then it gets longer and flips again. Here?s a small file that demonstrates the bad sawtooth waves: library(tuneR) sample_rate <- 12000 reverse <- FALSE mycolors=c("red","orange","yellow","green","cyan","blue","violet","magenta") plot(sawtooth(110,duration=round(sample_rate/100),samp.rate=sample_rate,xunit="samples")@left,typ...

Ccing the maintainer if the tuneR package. Looks to me like sawtooth (and square) don't behave as expected when using xunit="samples". Workaround is to use xunit="time" instead: sawtooth(110,duration=1/100,samp.rate=sample_rate,xunit="time") I looked at the code but found it to be opaque. -- Sent from my phone. Please excuse my brevity. On November 12, 2017 6:15:45 AM PST, Michael Tiemann <mdtiemann at gmail.com> wrote: >My tuneR sawtooth wave function generator is broken. > >When I use the...

...compile flac again, but i'm not sure about it... -- Miroslav Lichvar -------------- next part -------------- --- src/plugin_xmms/plugin.c.orig 2002-11-07 18:40:44.000000000 +0100 +++ src/plugin_xmms/plugin.c 2002-11-09 17:28:45.000000000 +0100 @@ -58,6 +58,7 @@ unsigned channels; unsigned sample_rate; unsigned length_in_msec; + gchar *title; AFormat sample_format; int seek_to_in_sec; FLAC__bool has_replaygain; @@ -114,6 +115,10 @@ #define SAMPLES_PER_WRITE 512 static FLAC__int32 reservoir_[FLAC__MAX_BLOCK_SIZE * 2/*for overflow*/ * FLAC_PLUGIN__MAX_SUPPORTED_CHANNELS]; static FLAC...

...ear end of files, here is better one. I'm sorry. -- Miroslav Lichvar -------------- next part -------------- --- src/plugin_xmms/plugin.c.orig 2002-11-07 18:40:44.000000000 +0100 +++ src/plugin_xmms/plugin.c 2002-11-10 00:40:20.000000000 +0100 @@ -58,6 +58,7 @@ unsigned channels; unsigned sample_rate; unsigned length_in_msec; + gchar *title; AFormat sample_format; int seek_to_in_sec; FLAC__bool has_replaygain; @@ -114,6 +115,10 @@ #define SAMPLES_PER_WRITE 512 static FLAC__int32 reservoir_[FLAC__MAX_BLOCK_SIZE * 2/*for overflow*/ * FLAC_PLUGIN__MAX_SUPPORTED_CHANNELS]; static FLAC...

severity 200435 normal thanks I received this bug report from a Debian user. I can't think of any reason offhand why the command line tool would work while the xmms plugin would fail. ----- Forwarded message from Stephan Niemz <st.n@gmx.net> ----- Date: Tue, 8 Jul 2003 10:24:57 +0200 From: Stephan Niemz <st.n@gmx.net> Resent-From: Stephan Niemz <st.n@gmx.net> To: Debian

Hi, Can anybody show me how to enable stereo AEC, I tried with the following code and the result is bad, degraded output. Init: ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); preprocess_state_left = speex_preprocess_state_init(frame_size, sample_rate); preprocess_state_right = speex_preprocess_state_init(frame_size, sample_rate); speex_preprocess_ctl(preprocess_state_left , SPEEX_PREPROCESS_SET_ECHO_STATE, ec_state ); speex_preprocess_ctl(preprocess_state_right ,...

...ot; static double rint(double x) { if (x < 0.0) return (double)(int)(x - 0.5); else return (double)(int)(x + 0.5); } theoraframes_info info; static int using_stdin = 0; typedef struct ff2theora{ AVFormatContext *context; int video_index; int audio_index; int deinterlace; int sample_rate; int channels; int disable_audio; float audio_quality; int output_width; int output_height; double fps; ImgReSampleContext *img_resample_ctx; /* for image resampling/resizing */ ReSampleContext *audio_resample_ctx; ogg_uint32_t aspect_numerator; ogg_uint32_t aspect_denominator; double...

...ic(and no speech), I do notice some attenuation, however, it does not suffer from any distortions. Following is the code I am using for initializing the opus multi-stream encoder: - <code> <defines and vars> #define CHANNELS 6 #define OPUS_SURROUND_MAPPING_FAMILY 1 #define SAMPLE_RATE 48000 typedef union t_SurroundInfo { unsigned char surroundInfo[1]; struct { unsigned char channels; unsigned char streams; unsigned char coupled_streams; unsigned char channel_mapping[MAX_SURROUND_CHANNELS]; }s; }SurroundInfo; </d...

...;get 8 bit (blocksize-1) from end of header", 0111 mean "get 16 bit (blocksize-1) from end of header") Why you don't use STRICT block size checking in FLAC__format_blocksize_is_subset() like this: FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate) { if(blocksize == 192 || blocksize == 576 || blocksize == 1152 || blocksize == 2304 || blocksize == 4608 || blocksize == 256 || blocksize == 512 || blocksize == 1024 || blocksize == 2048 || blocksize == 4096 || (sample_rate > 48000 && (blocksize == 8192 || blocksize == 16384)...

...-2178,10 +2178,15 @@ if(num_requested_seek_points < 0) { /*@@@@@@ workaround ogg bug: too many seekpoints makes table not fit in one page */ - if(e->use_ogg && e->total_samples_to_encode > 0 && e->total_samples_to_encode / e->sample_rate / 10 > 230) +#if FLAC__HAS_OGG + if(e->use_ogg && e->total_samples_to_encode > 0 && e->total_samples_to_encode / e->sample_rate / 10 > 230) { requested_seek_points = "230x;"; + } el...

I want the standard error associated with a correlation. I can calculate using cor & var, but am wondering if there are libraries that already provide this function. [[alternative HTML version deleted]]

...next packet, I'm trying to decode the packet with decode_fec = 1 and then the same packet with decode_fec = 0: In the code below, suggest to replace ‘ads->ch’ with ‘numChannels’ to make it more clear to what you refer to.) if(packet_lost ) { if(opus_packet_has_fec(buf, (opus_int32)len, sample_rate)) { fec_samples = opus_packet_get_samples_per_frame(buf, sample_rate); info("opus: there is fec packets=%d\n", fec_samples); n = opus_decode( ads->dec, buf, (opus_int32)len, sampv, fec_samples, 1); if (n < 0) { warning("opus: decode error: %s\n", opus_strer...

...;get 8 bit (blocksize-1) from end of header", 0111 mean "get 16 bit (blocksize-1) from end of header") Why you don't use STRICT block size checking in FLAC__format_blocksize_is_subset() like this: FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate) { if(blocksize == 192 || blocksize == 576 || blocksize == 1152 || blocksize == 2304 || blocksize == 4608 || blocksize == 256 || blocksize == 512 || blocksize == 1024 || blocksize == 2048 || blocksize == 4096 || (sample_rate > 48000 && (blocksize == 8192 || blocksize == 16384)...

...ic(and no speech), I do notice some attenuation, however, it does not suffer from any distortions. Following is the code I am using for initializing the opus multi-stream encoder: - <code> <defines and vars> #define CHANNELS 6 #define OPUS_SURROUND_MAPPING_FAMILY 1 #define SAMPLE_RATE 48000 typedef union t_SurroundInfo { unsigned char surroundInfo[1]; struct { unsigned char channels; unsigned char streams; unsigned char coupled_streams; unsigned char channel_mapping[MAX_SURROUND_CHANNELS]; }s; }SurroundInfo; </d...

On Fri, Nov 08, 2002 at 12:39:52PM -0800, Josh Coalson wrote: > --- Miroslav Lichvar <lichvarm@phoenix.inf.upol.cz> wrote: > > I have few notes: > > > > It seems there is changed API in CVS again. So, what about adding > > function like > > unsigned FLAC__format_frame_size(const FLAC__Frame *frame) > > which returns size of the frame in bytes. This

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

...lean get_file_info(char *filename, flac_file_info_struct *tmp_file_info) -{ - FLAC__StreamMetadata streaminfo; - - if(0 == filename) - filename = ""; +#include <gtk/gtk.h> - if(!FLAC__metadata_get_streaminfo(filename, &streaminfo)) { - return FALSE; - } - - tmp_file_info->sample_rate = streaminfo.data.stream_info.sample_rate; - tmp_file_info->channels = streaminfo.data.stream_info.channels; - tmp_file_info->bits_per_sample = streaminfo.data.stream_info.bits_per_sample; - tmp_file_info->total_samples = streaminfo.data.stream_info.total_samples; - - tmp_file_info->...

...;get 8 bit (blocksize-1) from end of header", 0111 mean "get 16 bit (blocksize-1) from end of header") Why you don't use STRICT block size checking in FLAC__format_blocksize_is_subset() like this: FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate) { if(blocksize == 192 || blocksize == 576 || blocksize == 1152 || blocksize == 2304 || blocksize == 4608 || blocksize == 256 || blocksize == 512 || blocksize == 1024 || blocksize == 2048 || blocksize == 4096 || (sample_rate > 48000 && (blocksize == 8192 || blocksize == 16384...

...; > Hi, > > Can anybody show me how to enable stereo AEC, I tried with the > following code and the result is bad, degraded output. > > Init: > > ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); > speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); > > preprocess_state_left = speex_preprocess_state_init(frame_size, > sample_rate); > preprocess_state_right = speex_preprocess_state_init(frame_size, > sample_rate); > > speex_preprocess_ctl(preprocess_state_left , > SPEEX_PREPROCESS_SET_ECHO_STATE, ec_state ); > sp...