mirror of
https://github.com/Rockbox/rockbox.git
synced 2025-11-22 19:42:40 -05:00
1) Always enable the DSP. 2) Change codec to output one 32-bit array per channel containing samples left-shifted to 28-bits (instead of 16-bit interleaved samples). 3) Remove the two 16KB internal predicterror_buffer arrays (we use the output arrays instead) 4) Small internal rearrangement of the code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7667 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman
parent
7da9477bc3
commit
f1844c4166
3 changed files with 420 additions and 411 deletions
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@ -31,6 +31,7 @@ extern char iramend[];
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#define destBufferSize (1024*16)
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char inputBuffer[1024*32]; /* Input buffer */
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int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE] IBSS_ATTR;
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size_t mdat_offset;
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struct codec_api* rb;
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struct codec_api* ci;
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@ -241,11 +242,10 @@ enum codec_status codec_start(struct codec_api* api)
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uint32_t elapsedtime;
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uint32_t sample_duration;
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uint32_t sample_byte_size;
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int outputBytes;
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int samplesdecoded;
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unsigned int i;
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unsigned char* buffer;
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alac_file alac;
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int16_t* pDestBuffer;
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/* Generic codec initialisation */
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TEST_CODEC_API(api);
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@ -261,9 +261,10 @@ enum codec_status codec_start(struct codec_api* api)
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(28));
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next_track:
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@ -275,12 +276,7 @@ enum codec_status codec_start(struct codec_api* api)
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while (!rb->taginfo_ready)
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rb->yield();
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if (rb->id3->frequency != NATIVE_FREQUENCY) {
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rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
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rb->configure(CODEC_DSP_ENABLE, (bool *)true);
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} else {
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rb->configure(CODEC_DSP_ENABLE, (bool *)false);
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}
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ci->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
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stream_create(&input_stream);
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@ -349,9 +345,8 @@ enum codec_status codec_start(struct codec_api* api)
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}
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/* Decode one block - returned samples will be host-endian */
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outputBytes = destBufferSize;
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rb->yield();
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pDestBuffer=decode_frame(&alac, buffer, &outputBytes, rb->yield);
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samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, rb->yield);
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/* Advance codec buffer - unless we did a read */
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if ((char*)buffer!=(char*)inputBuffer) {
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@ -360,7 +355,9 @@ enum codec_status codec_start(struct codec_api* api)
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/* Output the audio */
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rb->yield();
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while(!ci->pcmbuf_insert((char*)pDestBuffer,outputBytes))
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while(!ci->pcmbuf_insert_split(outputbuffer[0],
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outputbuffer[1],
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samplesdecoded*sizeof(int32_t)))
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rb->yield();
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/* Update the elapsed-time indicator */
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@ -51,10 +51,6 @@
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int16_t predictor_coef_table[32] IBSS_ATTR;
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int16_t predictor_coef_table_a[32] IBSS_ATTR;
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int16_t predictor_coef_table_b[32] IBSS_ATTR;
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int32_t predicterror_buffer_a[4096];
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int32_t predicterror_buffer_b[4096];
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int32_t outputsamples_buffer_a[4096] IBSS_ATTR;
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int32_t outputsamples_buffer_b[4096] IBSS_ATTR;
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void alac_set_info(alac_file *alac, char *inputbuffer)
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{
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@ -609,9 +605,9 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer,
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}
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}
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void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b,
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int16_t *buffer_out,
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int numchannels, int numsamples,
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void deinterlace_16(int32_t* buffer0,
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int32_t* buffer1,
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int numsamples,
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uint8_t interlacing_shift,
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uint8_t interlacing_leftweight)
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{
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@ -625,11 +621,13 @@ void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b,
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{
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int32_t difference, midright;
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midright = buffer_a[i];
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difference = buffer_b[i];
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midright = buffer0[i];
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difference = buffer1[i];
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buffer_out[i*numchannels] = (midright - ((difference * interlacing_leftweight) >> interlacing_shift)) + difference;
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buffer_out[i*numchannels + 1] = midright - ((difference * interlacing_leftweight) >> interlacing_shift);
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buffer0[i] = ((midright - ((difference * interlacing_leftweight)
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>> interlacing_shift)) + difference) << SCALE16;
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buffer1[i] = (midright - ((difference * interlacing_leftweight)
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>> interlacing_shift)) << SCALE16;
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}
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return;
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@ -638,39 +636,21 @@ void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b,
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/* otherwise basic interlacing took place */
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for (i = 0; i < numsamples; i++)
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{
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buffer_out[i*numchannels] = buffer_a[i];
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buffer_out[i*numchannels + 1] = buffer_b[i];
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buffer0[i] = buffer0[i] << SCALE16;
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buffer1[i] = buffer1[i] << SCALE16;
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}
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}
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int16_t* decode_frame(alac_file *alac,
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unsigned char *inbuffer,
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int *outputsize,
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static inline int decode_frame_mono(
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alac_file *alac,
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int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE],
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void (*yield)(void))
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{
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int channels;
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int16_t* outbuffer;
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int32_t outputsamples = alac->setinfo_max_samples_per_frame;
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/* setup the stream */
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alac->input_buffer = inbuffer;
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alac->input_buffer_bitaccumulator = 0;
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/* We can share the same buffer for outputbuffer
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and outputsamples_buffer_b - and hence have them both in IRAM*/
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outbuffer=(int16_t*)outputsamples_buffer_b;
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channels = readbits(alac, 3);
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*outputsize = outputsamples * alac->bytespersample;
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switch(channels)
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{
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case 0: /* 1 channel */
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{
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int hassize;
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int isnotcompressed;
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int readsamplesize;
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int outputsamples = alac->setinfo_max_samples_per_frame;
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int wasted_bytes;
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int ricemodifier;
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@ -694,7 +674,6 @@ int16_t* decode_frame(alac_file *alac,
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/* now read the number of samples,
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* as a 32bit integer */
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outputsamples = readbits(alac, 32);
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*outputsize = outputsamples * alac->bytespersample;
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}
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readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8);
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@ -734,7 +713,7 @@ int16_t* decode_frame(alac_file *alac,
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yield();
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basterdised_rice_decompress(alac,
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predicterror_buffer_a,
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outputbuffer[0],
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outputsamples,
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readsamplesize,
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alac->setinfo_rice_initialhistory,
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@ -746,8 +725,8 @@ int16_t* decode_frame(alac_file *alac,
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if (prediction_type == 0)
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{ /* adaptive fir */
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predictor_decompress_fir_adapt(predicterror_buffer_a,
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outputsamples_buffer_a,
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predictor_decompress_fir_adapt(outputbuffer[0],
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outputbuffer[0],
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outputsamples,
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readsamplesize,
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predictor_coef_table,
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@ -777,7 +756,7 @@ int16_t* decode_frame(alac_file *alac,
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audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
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outputsamples_buffer_a[i] = audiobits;
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outputbuffer[0][i] = audiobits;
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}
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}
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else
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@ -795,7 +774,7 @@ int16_t* decode_frame(alac_file *alac,
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audiobits |= readbits(alac, readsamplesize - 16);
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outputsamples_buffer_a[i] = audiobits;
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outputbuffer[0][i] = audiobits;
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}
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}
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/* wasted_bytes = 0; // unused */
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@ -811,8 +790,8 @@ int16_t* decode_frame(alac_file *alac,
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for (i = 0; i < outputsamples; i++)
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{
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/* Output mono data as stereo */
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outbuffer[i*2] = outputsamples_buffer_a[i];
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outbuffer[i*2+1] = outputsamples_buffer_a[i];
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outputbuffer[0][i] = outputbuffer[0][i] << SCALE16;
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outputbuffer[1][i] = outputbuffer[0][i];
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}
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break;
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}
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@ -824,14 +803,19 @@ int16_t* decode_frame(alac_file *alac,
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default:
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break;
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}
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break;
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return outputsamples;
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}
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case 1: /* 2 channels */
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static inline int decode_frame_stereo(
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alac_file *alac,
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int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE],
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void (*yield)(void))
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{
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int hassize;
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int isnotcompressed;
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int readsamplesize;
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int outputsamples = alac->setinfo_max_samples_per_frame;
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int wasted_bytes;
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uint8_t interlacing_shift;
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@ -855,7 +839,6 @@ int16_t* decode_frame(alac_file *alac,
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/* now read the number of samples,
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* as a 32bit integer */
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outputsamples = readbits(alac, 32);
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*outputsize = outputsamples * alac->bytespersample;
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}
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readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + 1;
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@ -913,7 +896,7 @@ int16_t* decode_frame(alac_file *alac,
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yield();
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/* channel 1 */
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basterdised_rice_decompress(alac,
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predicterror_buffer_a,
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outputbuffer[0],
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outputsamples,
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readsamplesize,
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alac->setinfo_rice_initialhistory,
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@ -924,8 +907,8 @@ int16_t* decode_frame(alac_file *alac,
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yield();
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if (prediction_type_a == 0)
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{ /* adaptive fir */
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predictor_decompress_fir_adapt(predicterror_buffer_a,
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outputsamples_buffer_a,
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predictor_decompress_fir_adapt(outputbuffer[0],
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outputbuffer[0],
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outputsamples,
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readsamplesize,
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predictor_coef_table_a,
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@ -941,7 +924,7 @@ int16_t* decode_frame(alac_file *alac,
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/* channel 2 */
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basterdised_rice_decompress(alac,
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predicterror_buffer_b,
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outputbuffer[1],
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outputsamples,
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readsamplesize,
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alac->setinfo_rice_initialhistory,
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@ -952,8 +935,8 @@ int16_t* decode_frame(alac_file *alac,
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yield();
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if (prediction_type_b == 0)
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{ /* adaptive fir */
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predictor_decompress_fir_adapt(predicterror_buffer_b,
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outputsamples_buffer_b,
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predictor_decompress_fir_adapt(outputbuffer[1],
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outputbuffer[1],
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outputsamples,
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readsamplesize,
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predictor_coef_table_b,
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@ -980,8 +963,8 @@ int16_t* decode_frame(alac_file *alac,
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audiobits_a = SIGN_EXTENDED32(audiobits_a, alac->setinfo_sample_size);
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audiobits_b = SIGN_EXTENDED32(audiobits_b, alac->setinfo_sample_size);
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outputsamples_buffer_a[i] = audiobits_a;
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outputsamples_buffer_b[i] = audiobits_b;
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outputbuffer[0][i] = audiobits_a;
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outputbuffer[1][i] = audiobits_b;
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}
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}
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else
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@ -1001,8 +984,8 @@ int16_t* decode_frame(alac_file *alac,
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audiobits_b = audiobits_b >> (32 - alac->setinfo_sample_size);
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audiobits_b |= readbits(alac, alac->setinfo_sample_size - 16);
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outputsamples_buffer_a[i] = audiobits_a;
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outputsamples_buffer_b[i] = audiobits_b;
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outputbuffer[0][i] = audiobits_a;
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outputbuffer[1][i] = audiobits_b;
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}
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}
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/* wasted_bytes = 0; */
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@ -1016,10 +999,8 @@ int16_t* decode_frame(alac_file *alac,
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{
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case 16:
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{
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deinterlace_16(outputsamples_buffer_a,
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outputsamples_buffer_b,
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(int16_t*)outbuffer,
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alac->numchannels,
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deinterlace_16(outputbuffer[0],
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outputbuffer[1],
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outputsamples,
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interlacing_shift,
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interlacing_leftweight);
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@ -1033,11 +1014,36 @@ int16_t* decode_frame(alac_file *alac,
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default:
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break;
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}
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return outputsamples;
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}
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int alac_decode_frame(alac_file *alac,
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unsigned char *inbuffer,
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int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE],
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void (*yield)(void))
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{
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int channels;
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int outputsamples;
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/* setup the stream */
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alac->input_buffer = inbuffer;
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alac->input_buffer_bitaccumulator = 0;
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channels = readbits(alac, 3);
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/* TODO: The mono and stereo functions should be combined. */
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switch(channels)
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{
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case 0: /* 1 channel */
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outputsamples=decode_frame_mono(alac,outputbuffer,yield);
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break;
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case 1: /* 2 channels */
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outputsamples=decode_frame_stereo(alac,outputbuffer,yield);
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break;
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default: /* Unsupported */
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return -1;
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}
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}
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return outbuffer;
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return outputsamples;
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}
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void create_alac(int samplesize, int numchannels, alac_file* alac)
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@ -1,6 +1,12 @@
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#ifndef __ALAC__DECOMP_H
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#define __ALAC__DECOMP_H
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/* Always output samples shifted to 28 bits */
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#define ALAC_OUTPUT_DEPTH 28
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#define SCALE16 (ALAC_OUTPUT_DEPTH - 16)
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#define ALAC_MAX_CHANNELS 2
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#define ALAC_BLOCKSIZE 4096 /* Number of samples per channel per block */
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typedef struct
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{
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unsigned char *input_buffer;
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@ -26,9 +32,9 @@ typedef struct
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} alac_file;
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void create_alac(int samplesize, int numchannels, alac_file* alac);
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int16_t* decode_frame(alac_file *alac,
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int alac_decode_frame(alac_file *alac,
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unsigned char *inbuffer,
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int *outputsize,
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int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE],
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void (*yield)(void));
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void alac_set_info(alac_file *alac, char *inputbuffer);
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