From 217c228738a92de03100412301e012fbe3d465ca Mon Sep 17 00:00:00 2001 From: Karl Blomster Date: Mon, 4 Aug 2008 07:13:41 +0000 Subject: [PATCH] misc small fixes in the ffmpeg audio provider, preparation for support of other sample formats than just 16-bit int Originally committed to SVN as r2296. --- aegisub/audio_provider_lavc.cpp | 31 +++++++++++++++++++++---------- 1 file changed, 21 insertions(+), 10 deletions(-) diff --git a/aegisub/audio_provider_lavc.cpp b/aegisub/audio_provider_lavc.cpp index b26be8654..5702e9842 100644 --- a/aegisub/audio_provider_lavc.cpp +++ b/aegisub/audio_provider_lavc.cpp @@ -112,16 +112,25 @@ LAVCAudioProvider::LAVCAudioProvider(Aegisub::String _filename) throw _T("ffmpeg audio provider: Failed to open audio decoder"); sample_rate = Options.AsInt(_T("Audio Sample Rate")); - if (!sample_rate) - sample_rate = codecContext->sample_rate; + if (!sample_rate) { + /* aegisub wants audio with sample rate higher than 32khz */ + if (codecContext->sample_rate < 32000) + sample_rate = 48000; + else + sample_rate = codecContext->sample_rate; + } - /* rely on the downmixing audio provider to do downmixing for us later */ + /* we rely on the intermediate audio provider to do downmixing for us later if necessary */ channels = codecContext->channels; - /* 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; - /* aegisub currently supports mono only, so always resample unless it's mono with the desired samplerate */ + /* FIXME: we need support for more audio types than just 16-bit int */ + switch (codecContext->sample_fmt) { + case SAMPLE_FMT_S16: bytes_per_sample = 2; break; + default: + throw _T("ffmpeg audio provider: Only 16-bit audio is supported"); + } + + /* initiate resampling if necessary */ if (sample_rate != codecContext->sample_rate) { rsct = audio_resample_init(channels, channels, sample_rate, codecContext->sample_rate); if (!rsct) @@ -137,7 +146,7 @@ LAVCAudioProvider::LAVCAudioProvider(Aegisub::String _filename) length = (double)lavcfile->fctx->duration / AV_TIME_BASE; else length = (double)stream->duration * av_q2d(stream->time_base); - num_samples = (int64_t)(length * sample_rate); // number of samples per channel + num_samples = (int64_t)(length * sample_rate); /* number of samples per channel */ buffer = (int16_t *)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE); if (!buffer) @@ -200,12 +209,13 @@ void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count) /* we're not dealing with video packets in this here provider */ if (packet.stream_index == audStream) { int size = packet.size; + uint8_t *data = packet.data; while (size > 0) { int temp_output_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; /* see constructor, it malloc()'s buffer to this */ int retval, decoded_bytes, decoded_samples; - retval = avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, packet.data, size); + 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 */ @@ -215,8 +225,9 @@ void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count) } decoded_bytes = temp_output_buffer_size; - decoded_samples = decoded_bytes / 2; /* 2 bytes per sample */ + decoded_samples = decoded_bytes / 2; /* FIXME: stop assuming everything is 16-bit! */ size -= retval; + data += retval; /* do we need to resample? */ if (rsct) {