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For #1506, refine the AAC samples guess algorithm.

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This commit is contained in:
winlin 2019-12-05 20:16:23 +08:00
parent 52c6c7c45d
commit 31f341e205
1 changed files with 21 additions and 7 deletions
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26
trunk/src/app/srs_app_hls.cpp
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@ -1230,17 +1230,31 @@ srs_error_t SrsHls::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* forma
aac_samples = 0; aac_samples = 0;
} }
// Use the diff to guess whether the samples is 1024 or 960. // The diff duration in ms between two FLV audio packets.
int nb_samples_per_frame = 1024; int diff = ::abs((int)(audio->timestamp - previous_audio_dts));
int diff = ::abs((int)(audio->timestamp - previous_audio_dts)) * srs_flv_srates[format->acodec->sound_rate];
previous_audio_dts = audio->timestamp; previous_audio_dts = audio->timestamp;
if (diff > 100 && diff < 950) {
nb_samples_per_frame = 960; // Guess the number of samples for each AAC frame.
// If samples is 1024, the sample-rate is 8000HZ, the diff should be 1024/8000s=128ms.
// If samples is 1024, the sample-rate is 44100HZ, the diff should be 1024/44100s=23ms.
// If samples is 2048, the sample-rate is 44100HZ, the diff should be 2048/44100s=46ms.
int nb_samples_per_frame = 0;
int guessNumberOfSamples = diff * srs_flv_srates[format->acodec->sound_rate] / 1000;
if (guessNumberOfSamples > 0) {
if (guessNumberOfSamples < 960) {
nb_samples_per_frame = 960;
} else if (guessNumberOfSamples < 1536) {
nb_samples_per_frame = 1024;
} else if (guessNumberOfSamples < 3072) {
nb_samples_per_frame = 2048;
} else {
nb_samples_per_frame = 4096;
}
} }
// Recalc the DTS by the samples of AAC. // Recalc the DTS by the samples of AAC.
int64_t dts = 90000 * aac_samples / srs_flv_srates[format->acodec->sound_rate];
aac_samples += nb_samples_per_frame; aac_samples += nb_samples_per_frame;
int64_t dts = 90000 * aac_samples / srs_flv_srates[format->acodec->sound_rate];
if ((err = controller->write_audio(format->audio, dts)) != srs_success) { if ((err = controller->write_audio(format->audio, dts)) != srs_success) {
return srs_error_wrap(err, "hls: write audio"); return srs_error_wrap(err, "hls: write audio");