test_codec: Fix some problems with writing WAV with DSP
On big-endian architecture, DSP output must be coverted to little- endian first. DSP output is also always interleaved stereo, 16 bit, NATIVE_FREQUENCY and wavinfo should be correct for this. Also, use standard clip_sample_16 already available. Change-Id: Ifa7b9fc77f0573070c7e79f059dc3000c437c42e
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@ -245,14 +245,6 @@ static int process_dsp(const void *ch1, const void *ch2, int count)
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return written_count;
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}
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static inline int32_t clip_sample(int32_t sample)
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{
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if ((int16_t)sample != sample)
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sample = 0x7fff ^ (sample >> 31);
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return sample;
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}
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/* Null output */
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static void pcmbuf_insert_null(const void *ch1, const void *ch2, int count)
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{
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@ -304,40 +296,39 @@ static int fill_buffer(int new_offset){
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/* WAV output or calculate crc32 of output*/
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static void pcmbuf_insert_wav_checksum(const void *ch1, const void *ch2, int count)
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{
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const int16_t* data1_16;
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const int16_t* data2_16;
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const int32_t* data1_32;
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const int32_t* data2_32;
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unsigned char* p = wavbuffer;
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const int scale = wavinfo.sampledepth - 15;
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const int dc_bias = 1 << (scale - 1);
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int channels = (wavinfo.stereomode == STEREO_MONO) ? 1 : 2;
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/* Prevent idle poweroff */
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rb->reset_poweroff_timer();
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if (use_dsp) {
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count = process_dsp(ch1, ch2, count);
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wavinfo.totalsamples += count;
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if (channels == 1)
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{
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unsigned char *s = dspbuffer, *d = dspbuffer;
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int c = count;
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while (c-- > 0)
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{
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*d++ = *s++;
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*d++ = *s++;
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s++;
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s++;
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}
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#ifdef ROCKBOX_BIG_ENDIAN
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unsigned char* p = dspbuffer;
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int i;
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for (i = 0; i < count; i++) {
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int2le16(p,*(int16_t *)p);
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p += 2;
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int2le16(p,*(int16_t *)p);
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p += 2;
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}
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#endif
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if (checksum) {
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crc32 = rb->crc_32(dspbuffer, count * 2 * sizeof (int16_t), crc32);
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} else {
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rb->write(wavinfo.fd, dspbuffer, count * 2 * sizeof (int16_t));
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}
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if (checksum)
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crc32 = rb->crc_32(dspbuffer, count * 2 * channels, crc32);
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else
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rb->write(wavinfo.fd, dspbuffer, count * 2 * channels);
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}
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else
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{
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const int16_t* data1_16;
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const int16_t* data2_16;
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const int32_t* data1_32;
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const int32_t* data2_32;
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unsigned char* p = wavbuffer;
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const int scale = wavinfo.sampledepth - 15;
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const int dc_bias = 1 << (scale - 1);
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if (wavinfo.sampledepth <= 16) {
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data1_16 = ch1;
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data2_16 = ch2;
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@ -378,18 +369,18 @@ static void pcmbuf_insert_wav_checksum(const void *ch1, const void *ch2, int cou
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{
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case STEREO_INTERLEAVED:
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while (count--) {
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int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
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int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
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p += 2;
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int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
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int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
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p += 2;
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}
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break;
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case STEREO_NONINTERLEAVED:
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while (count--) {
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int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
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int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
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p += 2;
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int2le16(p, clip_sample((*data2_32++ + dc_bias) >> scale));
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int2le16(p, clip_sample_16((*data2_32++ + dc_bias) >> scale));
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p += 2;
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}
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@ -397,7 +388,7 @@ static void pcmbuf_insert_wav_checksum(const void *ch1, const void *ch2, int cou
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case STEREO_MONO:
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while (count--) {
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int2le16(p, clip_sample((*data1_32++ + dc_bias) >> scale));
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int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
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p += 2;
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}
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break;
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@ -514,17 +505,17 @@ static void configure(int setting, intptr_t value)
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case DSP_SWITCH_FREQUENCY:
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case DSP_SET_FREQUENCY:
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DEBUGF("samplerate=%d\n",(int)value);
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wavinfo.samplerate = (int)value;
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wavinfo.samplerate = use_dsp ? NATIVE_FREQUENCY : (int)value;
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break;
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case DSP_SET_SAMPLE_DEPTH:
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DEBUGF("sampledepth = %d\n",(int)value);
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wavinfo.sampledepth=(int)value;
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wavinfo.sampledepth = use_dsp ? 16 : (int)value;
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break;
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case DSP_SET_STEREO_MODE:
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DEBUGF("Stereo mode = %d\n",(int)value);
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wavinfo.stereomode=(int)value;
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wavinfo.stereomode = use_dsp ? STEREO_INTERLEAVED : (int)value;
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break;
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}
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@ -698,8 +689,10 @@ static enum plugin_status test_track(const char* filename)
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ci.id3 = &track.id3;
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ci.curpos = 0;
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if (use_dsp)
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if (use_dsp) {
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rb->dsp_configure(ci.dsp, DSP_RESET, 0);
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rb->dsp_configure(ci.dsp, DSP_FLUSH, 0);
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}
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if (checksum)
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crc32 = 0xffffffff;
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