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Changed how the lavc audio decoder handles non-decodable audio packets. Previously, if decoding failed at any point in the audio stream, it would throw an exception that didn't get caught anywhere and hence crashed the program. Now, it'll just silently skip the broken packet and go on to next one, meaning that if the audio stream is actually corrupted and not just some weird stream with null packets you'll either get desyncs from the missing samples, or if the entire stream is non-decodable you'll just silence.

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tl;dr: Fixed loading of AAC files with the ffmpeg provider.

Originally committed to SVN as r2507.
This commit is contained in:
Karl Blomster 2008-12-17 06:22:05 +00:00
parent 04047e8f95
commit 6ad406446b
1 changed files with 9 additions and 8 deletions
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17
aegisub/audio_provider_lavc.cpp
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@ -223,18 +223,19 @@ void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
int retval, decoded_bytes, decoded_samples;
retval = avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, data, size);
if (retval <= 0)
throw _T("ffmpeg audio provider: failed to decode audio");
/* decoding succeeded but the output buffer is empty, go to next packet */
if (temp_output_buffer_size == 0) {
av_free_packet(&packet);
/* decoding failed, skip this packet and hope next one doesn't fail too */
if (retval < 0)
break;
/* throw _T("ffmpeg audio provider: failed to decode audio"); */
size -= retval;
data += retval;
/* decoding succeeded but this audio frame is empty, continue to next frame */
if (temp_output_buffer_size <= 0)
continue;
}
decoded_bytes = temp_output_buffer_size;
decoded_samples = decoded_bytes / bytes_per_sample; /* FIXME: stop assuming everything is 16-bit! */
size -= retval;
data += retval;
/* do we need to resample? */
if (rsct) {