search for: numchannel

I do the following to init flac: encoder = FLAC__stream_encoder_new(); e = FLAC__stream_encoder_set_do_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_channels (encoder, numChannels); e = FLAC__stream_encoder_set_bits_per_sample (encoder, jmin (24, bitsPerSample)); e = FLAC__stream_encoder_set_sample...

...type=VERBATIM ....... Any idea why/ where I have goofed? Thanks, James Code snippet: =================================================== FlacEncoder flacCompressor; bool setValue = false; // set up regular parameters setValue = flacCompressor.set_channels (numChannels); setValue = flacCompressor.set_bits_per_sample (bitsPerSample); setValue = flacCompressor.set_sample_rate (sampleRate); setValue = flacCompressor.set_blocksize(4608); setValue = flacCompressor.set_qlp_coeff_precision (0); setValue = flacCompressor.set_min_residual_partition_or...

...uint8_t *buffer8 = NULL; uint16_t *buffer16 = NULL; uint32_t *buffer32 = NULL; unsigned sample32; unsigned sample, channel; uint32_t bitsPerSample = this->get_bits_per_sample(); numFrames = inData.GetSize(); numChannels = this->get_channels(); // How big is our sample that we want to give to FLACC? // bitsPerSample is 8,16,24,32 // So 8 = no change for numFrames // 16 = half it // 24,32 = 1/4 the needs.. if (bitsPerSample == 16) numFrames = numFrames / 2; else...

--- Roland Rabien <Roland.Rabien@mackie.com> wrote: > I do the following to init flac: > > encoder = FLAC__stream_encoder_new(); > > e = FLAC__stream_encoder_set_do_mid_side_stereo (encoder, > numChannels == 2); > e = FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, > numChannels == 2); > e = FLAC__stream_encoder_set_channels (encoder, numChannels); > e = FLAC__stream_encoder_set_bits_per_sample (encoder, jmin > (24, > bitsPerSample)); >...

...us_decode(ads->dec, NULL, 0, sampv, (int)(*sampc/ads->ch), 0); and set the flag packet_lost=true; When I receive the 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, bu...

...ALITY, &tmp); tmp = 2; speex_encoder_ctl( enc_state, SPEEX_SET_COMPLEXITY, &tmp ); tmp = 1; speex_encoder_ctl( enc_state, SPEEX_SET_DTX, &tmp ); tmp = 1; speex_encoder_ctl( enc_state, SPEEX_SET_VAD, &tmp ); int numBytesEncoded = 0; int amountSamples = defaultfrequency * numchannels * 5; // frequency is 16000, channels = 1, 5 is for 5 seconds unsigned int total = 0; char *encoded = NULL; for( int j = 0; j < amountSamples/frame_size; j++ ) { // Encode the voice data speex_bits_reset(&bits); speex_encode_int(enc_state, (short*)((char*)p_raw1 + j*frame_size*...

...> Thanks, > > James > > > Code snippet: > =================================================== > FlacEncoder flacCompressor; > bool setValue = false; > > // set up regular parameters > setValue = flacCompressor.set_channels (numChannels); > setValue = flacCompressor.set_bits_per_sample (bitsPerSample); > setValue = flacCompressor.set_sample_rate (sampleRate); > setValue = flacCompressor.set_blocksize(4608); > setValue = flacCompressor.set_qlp_coeff_precision (0); > setValue = flacCompressor.set_...

...======================================================================== bool write (const int** samplesToWrite, int numSamples) { if (numSamples > 0) { float** const vorbisBuffer = vorbis_analysis_buffer (&vd, numSamples); for (int i = numChannels; --i >= 0;) { float* const dst = vorbisBuffer[i]; const int* const src = samplesToWrite [i]; if (src != 0 && dst != 0) { for (int j = 0; j < numSamples; ++j)...

...======================================================================== bool write (const int** samplesToWrite, int numSamples) { if (numSamples > 0) { float** const vorbisBuffer = vorbis_analysis_buffer (&vd, numSamples); for (int i = numChannels; --i >= 0;) { float* const dst = vorbisBuffer[i]; const int* const src = samplesToWrite [i]; if (src != 0 && dst != 0) { for (int j = 0; j < numSamples; ++j)...

I am deeply sorry about the corrupt message just being sent; there seems to have been a compatibility issue with my mailer and my browser. This is an identical copy of the previous message: Hello Martin, Vorbis encoders are lossy, which is in a sense equivalent to converting the sample size of the raw PCM stream into something that would result in the desired bitrate. The "sample size"

...fically, does the pcm total only account for one channel? How are channels represented in the file? Also, Is ov_read generating different amounts of data based on the word parameter? I can't think of any other reason why my calculation for a reasonable uncompressed buffer size is ov_pcm_total * numChannels * (the word size I pass to ov_read). Thanks in advance, Blake

...sis: %d %p %p\n", v_rv0, vb, op); if (v_rv0 == 0) { /* Block is valid for synthesis and synthesis is ready */ v_rv0 = vorbis_synthesis_blockin(vd, vb); if (v_rv0 == 0) { /* Synthesis was successful. Pull out the data */ int pcmSamples = 1; ogg_int32_t **pcm; int numChannels = 2; /* FIXME: Should really pull this from info */ int ii, jj; while(pcmSamples > 0) { pcmSamples = vorbis_synthesis_pcmout(vd, &pcm); /* Interleaver for output PCM */ for(ii=0; ii<pcmSamples; ++ii) { for(jj=0; jj<numChannels; ++jj) { ogg_int32_t...