Probably attaching the file is more readable.
Thanks for your help.
On Thu, Aug 14, 2014 at 2:12 PM, Jose Pablo Carballo
<jose.carba...@ridgerun.com> wrote:
> On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.m...@gmail.com> wrote:
>> Jose Pablo Carballo <jose.carba...@ridgerun.com> wrote:
>>
>>> - channels = 2;
>>> - bps = 16;
>>> + channels = ((unsigned)buffer[23] << 8) | buffer[22];
>>> + bps = ((unsigned)buffer[35] << 8) | buffer[34];
>>> total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8)
>>> | buffer[41]) << 8) | buffer[40]) / 4;
>>>
>>
>> I suspect that the expression for total_samples should be not
>>
>> (.....) / 4
>>
>> but
>>
>> (.....) / (channels * bps/8)
>
> Yes, that's correct.
>
> Here is the final diff I used to use the example to convert the 24bps file:
>
> diff --git a/examples/c/encode/file/main.c b/examples/c/encode/file/main.c
> index e3bdea8..b1cf374 100644
> --- a/examples/c/encode/file/main.c
> +++ b/examples/c/encode/file/main.c
> @@ -40,8 +40,10 @@ static void progress_callback(const
> FLAC__StreamEncoder *encoder, FLAC__uint64 b
>
> #define READSIZE 1024
>
> +#define BPS 24 /* Bits per sample */
> +
> static unsigned total_samples = 0; /* can use a 32-bit number due to
> WAVE size limitations */
> -static FLAC__byte buffer[READSIZE/*samples*/ * 2/*bytes_per_sample*/
> * 2/*channels*/]; /* we read the WAVE data into here */
> +static FLAC__byte buffer[READSIZE/*samples*/ * BPS/8
> /*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into
> here */
> static FLAC__int32 pcm[READSIZE/*samples*/ * 2/*channels*/];
>
> int main(int argc, char *argv[])
> @@ -73,14 +75,18 @@ int main(int argc, char *argv[])
> memcmp(buffer+8, "WAVEfmt \020\000\000\000\001\000\002\000", 16) ||
> memcmp(buffer+32, "\004\000\020\000data", 8)
> ) {
> +#if BPS == 16
> fprintf(stderr, "ERROR: invalid/unsupported WAVE file, only 16bps
> stereo WAVE in canonical form allowed\n");
> fclose(fin);
> return 1;
> +#elif BPS == 24
> + /* TODO: check wav header for 24bps */
> +#endif
> }
> sample_rate = ((((((unsigned)buffer[27] << 8) | buffer[26]) << 8) |
> buffer[25]) << 8) | buffer[24];
> - channels = 2;
> - bps = 16;
> - total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8)
> | buffer[41]) << 8) | buffer[40]) / 4;
> + channels = ((unsigned)buffer[23] << 8) | buffer[22];
> + bps = ((unsigned)buffer[35] << 8) | buffer[34];
> + total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8)
> | buffer[41]) << 8) | buffer[40]) / (channels * bps/8);
>
> /* allocate the encoder */
> if((encoder = FLAC__stream_encoder_new()) == NULL) {
> @@ -89,7 +95,12 @@ int main(int argc, char *argv[])
> return 1;
> }
>
> - ok &= FLAC__stream_encoder_set_verify(encoder, true);
> + if (bps == 16) {
> + /* TODO: Understand why verify doesn't work for 24bps - fails with
> + * FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA when calling
> + * FLAC__stream_encoder_process_interleaved().*/
> + ok &= FLAC__stream_encoder_set_verify(encoder, true);
> + }
> ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
> ok &= FLAC__stream_encoder_set_channels(encoder, channels);
> ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
> @@ -138,11 +149,23 @@ int main(int argc, char *argv[])
> /* convert the packed little-endian 16-bit PCM samples from WAVE
> into an interleaved FLAC__int32 buffer for libFLAC */
> size_t i;
> for(i = 0; i < need*channels; i++) {
> - /* inefficient but simple and works on big- or little-endian machines */
> - pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8)
> | (FLAC__int16)buffer[2*i]);
> + if (bps == 16) {
> + /* inefficient but simple and works on big- or little-endian machines */
> + pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8)
> | (FLAC__int16)buffer[2*i]);
> + } else if (bps == 24) {
> + pcm[i] = (FLAC__int32)buffer[3*i+2];
> + pcm[i] <<= 8;
> + pcm[i] |= (FLAC__int32)buffer[3*i+1];
> + pcm[i] <<= 8;
> + pcm[i] |= (FLAC__int32)buffer[3*i];
> + }
> }
> /* feed samples to encoder */
> ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need);
> + if (!ok) {
> + fprintf(stderr, "encoding: FLAC__stream_encoder_process_interleaved
> FAILED");
> + fprintf(stderr, " state: %s\n",
> FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
> + }
> }
> left -= need;
> }
diff --git a/examples/c/encode/file/main.c b/examples/c/encode/file/main.c
index e3bdea8..b1cf374 100644
--- a/examples/c/encode/file/main.c
+++ b/examples/c/encode/file/main.c
@@ -40,8 +40,10 @@ static void progress_callback(const FLAC__StreamEncoder *encoder, FLAC__uint64 b
#define READSIZE 1024
+#define BPS 24 /* Bits per sample */
+
static unsigned total_samples = 0; /* can use a 32-bit number due to WAVE size limitations */
-static FLAC__byte buffer[READSIZE/*samples*/ * 2/*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */
+static FLAC__byte buffer[READSIZE/*samples*/ * BPS/8 /*bytes_per_sample*/ * 2/*channels*/]; /* we read the WAVE data into here */
static FLAC__int32 pcm[READSIZE/*samples*/ * 2/*channels*/];
int main(int argc, char *argv[])
@@ -73,14 +75,18 @@ int main(int argc, char *argv[])
memcmp(buffer+8, "WAVEfmt \020\000\000\000\001\000\002\000", 16) ||
memcmp(buffer+32, "\004\000\020\000data", 8)
) {
+#if BPS == 16
fprintf(stderr, "ERROR: invalid/unsupported WAVE file, only 16bps stereo WAVE in canonical form allowed\n");
fclose(fin);
return 1;
+#elif BPS == 24
+ /* TODO: check wav header for 24bps */
+#endif
}
sample_rate = ((((((unsigned)buffer[27] << 8) | buffer[26]) << 8) | buffer[25]) << 8) | buffer[24];
- channels = 2;
- bps = 16;
- total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8) | buffer[41]) << 8) | buffer[40]) / 4;
+ channels = ((unsigned)buffer[23] << 8) | buffer[22];
+ bps = ((unsigned)buffer[35] << 8) | buffer[34];
+ total_samples = (((((((unsigned)buffer[43] << 8) | buffer[42]) << 8) | buffer[41]) << 8) | buffer[40]) / (channels * bps/8);
/* allocate the encoder */
if((encoder = FLAC__stream_encoder_new()) == NULL) {
@@ -89,7 +95,12 @@ int main(int argc, char *argv[])
return 1;
}
- ok &= FLAC__stream_encoder_set_verify(encoder, true);
+ if (bps == 16) {
+ /* TODO: Understand why verify doesn't work for 24bps - fails with
+ * FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA when calling
+ * FLAC__stream_encoder_process_interleaved().*/
+ ok &= FLAC__stream_encoder_set_verify(encoder, true);
+ }
ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
ok &= FLAC__stream_encoder_set_channels(encoder, channels);
ok &= FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
@@ -138,11 +149,23 @@ int main(int argc, char *argv[])
/* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
size_t i;
for(i = 0; i < need*channels; i++) {
- /* inefficient but simple and works on big- or little-endian machines */
- pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
+ if (bps == 16) {
+ /* inefficient but simple and works on big- or little-endian machines */
+ pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
+ } else if (bps == 24) {
+ pcm[i] = (FLAC__int32)buffer[3*i+2];
+ pcm[i] <<= 8;
+ pcm[i] |= (FLAC__int32)buffer[3*i+1];
+ pcm[i] <<= 8;
+ pcm[i] |= (FLAC__int32)buffer[3*i];
+ }
}
/* feed samples to encoder */
ok = FLAC__stream_encoder_process_interleaved(encoder, pcm, need);
+ if (!ok) {
+ fprintf(stderr, "encoding: FLAC__stream_encoder_process_interleaved FAILED");
+ fprintf(stderr, " state: %s\n", FLAC__StreamEncoderStateString[FLAC__stream_encoder_get_state(encoder)]);
+ }
}
left -= need;
}
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