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* decoder: use audio_format_init_checked()Max Kellermann2009-11-1415-85/+124
| | | | | | Let the audio_check library verify the audio format in all (relevant, i.e. non-hardcoded) plugins.
* audio_check: checker functions for audio_format attributesMax Kellermann2009-11-144-12/+138
| | | | | These functions are a wrapper for audio_valid_X(). On error, they return a GError object.
* decoder/sidplay: correctly calculate floating point timeMax Kellermann2009-11-141-8/+11
| | | | | Internally, use only the integer time. When needed, convert it to a floating point seconds value.
* player_thread: corrected two assertions on "queued"Max Kellermann2009-11-141-2/+2
| | | | At this point, the function may be called from the SEEK handler.
* player_thread: initialize chunk->times in silence generatorMax Kellermann2009-11-122-1/+5
| | | | | | | | | | When waiting for the decoder to provide more data, the player thread generates silence chunks if needed. However, it forgot to initialize the chunk.times attribute, which had now an undefined value. This patch sets it to -1.0, meaning "value is undefined". Add a ">= 0.0" check to audio_output_all_check(). This fixes spurious relative seeking errors, because sometimes, the "elapsed" value falls back to 0.0.
* player_control: hold lock while reading statusMax Kellermann2009-11-121-1/+4
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* added .#* to .gitignoreMax Kellermann2009-11-122-1/+1
| | | | Temporary editor files.
* include config.h in all sourcesMax Kellermann2009-11-12212-65/+329
| | | | | | After we've been hit by Large File Support problems several times in the past week (which only occur on 32 bit platforms, which I don't have), this is yet another attempt to fix the issue.
* decoder/vorbis: fixed gcc "signed" warningMax Kellermann2009-11-121-2/+2
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* directory: include config.hMax Kellermann2009-11-111-0/+1
| | | | | *sigh* another Large File breakage. ino_t/dev_t this time. We need to include config.h in directory.h to get this straight.
* decoder/wavpack: allow more than 2 channelsMax Kellermann2009-11-112-3/+4
| | | | | Remove the OPEN_2CH_MAX option. MPD's support for surround sound is still clunky, but we're working on it.
* decoder/wavpack: activate 32 bit supportMax Kellermann2009-11-112-13/+8
| | | | | | | | | | MPD has been supporting 32 bit samples since version 0.15. This patch changes one check, and removes the 32->24 conversion code. Note that WavPack floating point samples have 32 bits, and MPD doesn't have a special check for floating point - therefore, this WavPack plugin still returns 24 bit integer samples as before (until we have float support in the MPD core).
* decoder/vorbis: initialize before entering the loopMax Kellermann2009-11-111-21/+37
| | | | | | | Call decoder_initialize() before entering the loop. We don't need to call ov_read() before ov_info(). When the stream number changes, check if the audio format is still the same.
* decoder/vorbis: moved error strings to vorbis_strerror()Max Kellermann2009-11-111-24/+26
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* decoder/vorbis: removed the OggCallbackData typedefMax Kellermann2009-11-111-6/+7
| | | | Use the struct name instead.
* decoder/vorbis: fix typo in commentMax Kellermann2009-11-111-1/+1
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* decoder/vorbis: removed redundant "bits" initializationMax Kellermann2009-11-111-1/+0
| | | | This is done by audio_format_init().
* decoder/flac: check "seekable" in libFLAC callbacksMax Kellermann2009-11-111-0/+6
| | | | | Return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED if this input stream does not support seeking.
* decoder/flac: moved code to flac_data_get_audio_format()Max Kellermann2009-11-114-32/+51
| | | | | | Remove the audio_format attribute, add "frame_size" instead. The audio_format initialization and check is moved both to flac_data_get_audio_format().
* decoder/flac: use stream_info instead of audio_formatMax Kellermann2009-11-112-4/+4
| | | | | Use the sample rate stored in the stream_info struct instead of the audio_format struct.
* decoder/flac: use frame header instead of audio_formatMax Kellermann2009-11-111-3/+3
| | | | | | When calculating the properties of the frame, use sample_rate and other information from the frame header instead of the stored audio_format object.
* decoder/oggflac: moved stream_info check to oggflac_decode()Max Kellermann2009-11-111-6/+5
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* decoder/flac: calculate time stamp from current frameMax Kellermann2009-11-114-17/+17
| | | | | | | | | Don't update a float timestamp, this will make imprecisions add up after a while. We already have the number of the current frame, let's just calculate the float timestamp from that for every decoder_data() command. For this, we need to add the attribute "first_frame", for CUE sheet songs.
* decoder/flac: calculate bit rate in flac_common_write()Max Kellermann2009-11-114-17/+25
| | | | | | Removed the "bit_rate" attribute from the flac_data struct. Pass the number of bytes since the last call to flac_common_write(), and let it calculate the bit rate.
* decoder/flac: store the whole stream info object, not durationMax Kellermann2009-11-114-7/+36
| | | | | | | We don't want to work with floating point values if possible. Get the integer number of frames from the FLAC__StreamMetadata_StreamInfo object, and convert it into a float duration on demand. This patch adds a check if the STREAMINFO packet has been received yet.
* decoder/flac: merge code into flac_decoder_initialize()Max Kellermann2009-11-111-50/+39
| | | | | Wrapper for FLAC__stream_decoder_process_until_end_of_metadata(), decoder_initialized().
* decoder/flac: merged code into flac_decoder_new()Max Kellermann2009-11-111-28/+27
| | | | | Convenience wrapper for FLAC__stream_decoder_new() and FLAC__stream_decoder_set_metadata_respond().
* decoder/flac: free the "pathname" variable earlierMax Kellermann2009-11-111-31/+15
| | | | | Free the pointer right after its last use, i.e. after the FLAC__stream_decoder_init_file() call.
* decoder/flac: emulate FLAC__stream_decoder_init_stream()Max Kellermann2009-11-112-30/+44
| | | | Remove the wrapper flac_init().
* decoder/flac: use the new API functionsMax Kellermann2009-11-112-124/+89
| | | | | Use the type and function names of the libFLAC 1.1.3 API. Map the new API to the old one with macros.
* decoder/flac: removed the fake flac_ogg_init() fallbackMax Kellermann2009-11-112-2/+4
| | | | Don't even try to call it with an old libFLAC API.
* decoder/flac: moved code to flac_compat.hMax Kellermann2009-11-114-113/+135
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* decoder/{flac,vorbis}: include config.h for LFSMax Kellermann2009-11-113-0/+3
| | | | Allow those plugins to open large files on 32 bit platforms.
* decoder/flac: merged code into flac_decoder_loop()Max Kellermann2009-11-111-101/+55
| | | | | | | The decoder loop of flac_decode_internal(), flac_container_decode() and flac_filedecode_internal() is merged into this one function. This unifies the code, and uses the frame number to identify the end of a CUE sub song.
* decoder/flac: keep track of current frame numberMax Kellermann2009-11-114-0/+12
| | | | We need this for more exact end-of-subsong detection for CUE files.
* Merge remote branch 'origin/v0.15.x'Max Kellermann2009-11-113-14/+18
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| * decoder/flac: fixed CUE seeking range checkMax Kellermann2009-11-112-14/+9
| | | | | | | | | | | | If flac_container_decode() gets a seek destination which is out of range, it ignores the SEEK command (never finishes it). This leads to MPD lockup, because the player thread waits for completion.
| * oggflac: rewind stream after FLAC detectionMax Kellermann2009-11-112-0/+9
| | | | | | | | | | The oggflac plugin has been completely broken for quite a while and nobody has noticed - maybe we should remove it?
* | fd_util: don't call fd_set_nonblock() if open() has failedMax Kellermann2009-11-111-1/+2
| | | | | | | | This fixes an assertion failure.
* | added missing config.h includes for extended LFS supportMax Kellermann2009-11-117-4/+8
| | | | | | | | | | All sources which might work with large files must include config.h, to get Large File Support on 32 bit platforms.
* | update: added missing config.h includesMax Kellermann2009-11-112-1/+7
| | | | | | | | This broke sticker and archive support.
* | decoder/flac: removed redundant NULL checksMax Kellermann2009-11-111-9/+3
| | | | | | | | After the decoder loop, "flac_dec" is always set.
* | decoder/flac: moved code to flac_pcm.cMax Kellermann2009-11-114-81/+135
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* | decoder/flac: moved code to flac_metadata.cMax Kellermann2009-11-117-176/+244
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* | decoder/flac: return replay_gain_info object from helper functionMax Kellermann2009-11-111-28/+24
| | | | | | | | | | Make the function more generic by not passing "struct flac_data" to it.
* | Merge branch 'master' of git://git.musicpd.org/metyl/mpdMax Kellermann2009-11-114-1/+268
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| * | wave_encoder: new encoder for streaming PCM wave files.Viliam Mateicka2009-11-104-1/+268
| | | | | | | | | | | | | | | | | | When using wave encoder with httpd audio output mpd can input this stream via http and audiofile decoder. This for example opens simple way to configure lossless audio streaming port(like jack or pulseaudio does but without overhead). Another possibility can be using it for gathering raw data for visualization plugins (If sync issue will be resolved)
* | | decoder/flac: merged some code into flac_tag_apply_metadata()Max Kellermann2009-11-114-25/+27
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* | | decoder/oggflac: initialize the "tag" variableMax Kellermann2009-11-101-1/+2
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* | | decoder/flac: don't use float to calculate song durationMax Kellermann2009-11-103-8/+10
| | | | | | | | | | | | | | | Simple (up-rounding) integer division is good enough. We're casting the result back to an integer anyway.
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