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cleanup of the lavc audio provider; renamed some variables, added some comments and restructured a bit. as an added improvement it will now no longer resample unless strictly necessary. 

Originally committed to SVN as r2093.
This commit is contained in:
Karl Blomster 2008-03-21 19:52:14 +00:00
parent b060751cfe
commit 6d8f862aed
1 changed files with 41 additions and 31 deletions
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70
aegisub/audio_provider_lavc.cpp
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@ -105,20 +105,23 @@ LAVCAudioProvider::LAVCAudioProvider(Aegisub::String _filename)
if (avcodec_open(codecContext, codec) < 0) if (avcodec_open(codecContext, codec) < 0)
throw _T("Failed to open audio decoder"); throw _T("Failed to open audio decoder");
/* aegisub currently supports mono only, so always resample */
sample_rate = Options.AsInt(_T("Audio Sample Rate")); sample_rate = Options.AsInt(_T("Audio Sample Rate"));
if (!sample_rate) if (!sample_rate)
sample_rate = codecContext->sample_rate; sample_rate = codecContext->sample_rate;
channels = 1; channels = 1;
/* FIXME: this entire provider always assumes 16-bit audio. Currently that isn't a problem since
ffmpeg always converts everything to 16-bit, but in the future it might become one. */
bytes_per_sample = 2; bytes_per_sample = 2;
rsct = audio_resample_init(1, codecContext->channels, sample_rate, codecContext->sample_rate); /* aegisub currently supports mono only, so always resample unless it's mono with the desired samplerate */
if (!rsct) if ((sample_rate != codecContext->sample_rate) || (codecContext->channels > 1)) {
throw _T("Failed to initialize resampling"); rsct = audio_resample_init(1, codecContext->channels, sample_rate, codecContext->sample_rate);
if (!rsct)
throw _T("Failed to initialize resampling");
resample_ratio = (float)sample_rate / (float)codecContext->sample_rate; resample_ratio = (float)sample_rate / (float)codecContext->sample_rate;
}
double length = (double)stream->duration * av_q2d(stream->time_base); double length = (double)stream->duration * av_q2d(stream->time_base);
num_samples = (int64_t)(length * sample_rate); num_samples = (int64_t)(length * sample_rate);
@ -154,42 +157,49 @@ void LAVCAudioProvider::Destroy()
void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count) void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
{ {
int16_t *_buf = (int16_t *)buf; int16_t *_buf = (int16_t *)buf;
int64_t _count = num_samples - start; int64_t samples_to_decode = num_samples - start; /* samples left to the end of the stream */
if (count < _count) if (count < samples_to_decode) /* haven't reached the end yet, so just decode the requested number of samples */
_count = count; samples_to_decode = count;
if (_count < 0) if (samples_to_decode < 0) /* requested beyond the end of the stream */
_count = 0; samples_to_decode = 0;
memset(_buf + _count, 0, (count - _count) << 1); /* if we got asked for more samples than there are left in the stream, add zeros to the decoding buffer until
we have enough to fill the request */
memset(_buf + samples_to_decode, 0, (count - samples_to_decode) * 2);
AVPacket packet; AVPacket packet;
while (_count > 0 && av_read_frame(lavcfile->fctx, &packet) >= 0) { while (samples_to_decode > 0 && av_read_frame(lavcfile->fctx, &packet) >= 0) {
while (packet.stream_index == audStream) { while (packet.stream_index == audStream) {
int bytesout = AVCODEC_MAX_AUDIO_FRAME_SIZE; /* see constructor, it malloc()'s buffer to this */ int temp_output_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; /* see constructor, it malloc()'s buffer to this */
int samples; int decoded_samples;
/* returns negative if error, 0 if no frame decoded, number of bytes used on success */
if (avcodec_decode_audio2(codecContext, buffer, &bytesout, packet.data, packet.size) <= 0) if (avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, packet.data, packet.size) <= 0)
throw _T("Failed to decode audio"); throw _T("Failed to decode audio");
if (bytesout == 0) /* sanity checking */ if (temp_output_buffer_size == 0) /* gets changed to number of bytes actually output, so this is sanity checking */
break; break;
samples = bytesout >> 1; decoded_samples = temp_output_buffer_size / 2;
/* do we need to resample? */
if (rsct) { if (rsct) {
if ((int64_t)(samples * resample_ratio / codecContext->channels) > _count) if ((int64_t)(decoded_samples * resample_ratio / codecContext->channels) > samples_to_decode)
samples = (int64_t)(_count / resample_ratio * codecContext->channels); decoded_samples = (int64_t)(samples_to_decode / resample_ratio * codecContext->channels);
samples = audio_resample(rsct, _buf, buffer, samples / codecContext->channels); decoded_samples = audio_resample(rsct, _buf, buffer, decoded_samples / codecContext->channels);
assert(samples <= _count); /* make sure we somehow didn't end up with more samples than we wanted */
assert(decoded_samples <= samples_to_decode);
} else { } else {
/* currently dead code, rsct != NULL because we're resampling for mono */ /* no resampling needed, just copy to the buffer */
if (samples > _count) /* if (decoded_samples > samples_to_decode)
samples = _count; decoded_samples = samples_to_decode; */
memcpy(_buf, buffer, samples << 1); /* I do not understand the point of the above, changed to a more reasonable assertation instead -Fluff */
assert(decoded_samples <= samples_to_decode);
memcpy(_buf, buffer, temp_output_buffer_size);
} }
_buf += samples; _buf += decoded_samples;
_count -= samples; samples_to_decode -= decoded_samples;
break; /* break; */ /* why did this loop need to be broken manually? */
} }
av_free_packet(&packet); av_free_packet(&packet);