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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-04-01 10:15:29 +0200 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-04-01 10:15:29 +0200 |
commit | c5cbfc7e8feb92a65a825698f6c9ca7f6df652db (patch) | |
tree | b602c4897ef86dce2d0d4de633aea7ac2d5739da /src/dabplus-enc.cpp | |
parent | b04e8d19d684479b4a5c096789ead96f4b1a60ee (diff) | |
download | ODR-AudioEnc-c5cbfc7e8feb92a65a825698f6c9ca7f6df652db.tar.gz ODR-AudioEnc-c5cbfc7e8feb92a65a825698f6c9ca7f6df652db.tar.bz2 ODR-AudioEnc-c5cbfc7e8feb92a65a825698f6c9ca7f6df652db.zip |
Fix libtoolame drift compensation
Diffstat (limited to 'src/dabplus-enc.cpp')
-rw-r--r-- | src/dabplus-enc.cpp | 42 |
1 files changed, 36 insertions, 6 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp index 5b2df47..34add1c 100644 --- a/src/dabplus-enc.cpp +++ b/src/dabplus-enc.cpp @@ -688,12 +688,11 @@ int main(int argc, char *argv[]) /* We assume that we need to call the encoder * enc_calls_per_output before it gives us one encoded audio * frame. This information is used when the alsa drift compensation - * is active + * is active. This is only valid for FDK-AAC. */ - const int enc_calls_per_output = - (selected_encoder == encoder_selection_t::fdk_dabplus) ? - ((aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000) : - 1; + const int enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ? + sample_rate / 8000 : + sample_rate / 16000; int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL; @@ -819,7 +818,8 @@ int main(int argc, char *argv[]) AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; // -------------- wait the right amount of time - if (drift_compensation || jack_name) { + if ( (drift_compensation or jack_name) and + selected_encoder == encoder_selection_t::fdk_dabplus) { struct timespec tp_now; clock_gettime(CLOCK_MONOTONIC, &tp_now); @@ -1088,6 +1088,36 @@ int main(int argc, char *argv[]) else { numOutBytes = toolame_finish(&outbuf[0], outbuf.size()); } + + // We throttle Toolame by measuring the incoming audio size, because we + // know it always eats 1152 samples, regardless of configuration. + if (read_bytes and (drift_compensation or jack_name)) { + struct timespec tp_now; + clock_gettime(CLOCK_MONOTONIC, &tp_now); + + unsigned long time_now = (1000000000ul * tp_now.tv_sec) + + tp_now.tv_nsec; + unsigned long time_next = (1000000000ul * tp_next.tv_sec) + + tp_next.tv_nsec; + + // wait_time is in nanoseconds + const unsigned long wait_time = 1000000000ul * + 1152ul / (unsigned long)sample_rate; + + unsigned long waiting = wait_time - (time_now - time_next); + if ((time_now - time_next) < wait_time) { + //printf("Sleep %zuus\n", waiting / 1000); + usleep(waiting / 1000); + } + + // Move our time_counter into the future, for + // the next frame. + tp_next.tv_nsec += wait_time; + if (tp_next.tv_nsec > 1000000000L) { + tp_next.tv_nsec -= 1000000000L; + tp_next.tv_sec += 1; + } + } } /* Check if the encoder has generated output data */ |