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* ffmpeg: moved code to ffmpeg_send_frame()Max Kellermann2008-10-301-27/+34
| | | | | Move code from ffmpeg_decode_internal() to make it smaller and more readable.
* ffmpeg: report seek errors to MPDMax Kellermann2008-10-301-5/+4
| | | | | The decoder API provides the function decoder_seek_error() to report seek errors. Use this function instead of logging the error.
* ffmpeg: simplified mpdurl_read()Max Kellermann2008-10-301-13/+11
| | | | | The function mpdurl_read() is too complicated, and uses the wrong data types.
* ffmpeg: call tag_free() instead of free()Max Kellermann2008-10-301-1/+1
| | | | | tag objects must be freed with tag_free() to ensure that all resources are freed.
* ffmpeg: eliminated local variable "tag"Max Kellermann2008-10-301-7/+4
| | | | | The function ffmpeg_tag() already has the variable base.tag, which can be used for this.
* ffmpeg: make ffmpeg_helper() return boolMax Kellermann2008-10-301-24/+21
| | | | | ffmpeg_try_decode() did not interpret ffmpeg_helper()'s return value properly; migrate everything to bool to make it consistent.
* ffmpeg: removed debug messagesMax Kellermann2008-10-301-35/+5
| | | | We don't need those anymore, they just fill the log.
* ffmpeg: initialize base.decoderMax Kellermann2008-10-301-0/+1
| | | | | ffmpeg_tag() did not initialize base.decoder, which made valgrind unhappy, and can lead to a egmentation fault.
* listen: set file mode 666 on the unix socketMax Kellermann2008-10-301-0/+4
| | | | | | Depending on MPD's umask, the file permissions of the unix socket were too restrictive, and many clients were not able to connect. Do a chmod(0666) on the socket, to allow everybody to connect.
* pulse: fix segfault without sink nameMax Kellermann2008-10-301-2/+2
| | | | | Commit d692e9 broke configurations without explicit server or sink configuration. Check for getBlockParam()==NULL.
* mpc: provide 24 bit samplesMax Kellermann2008-10-301-7/+6
| | | | | | Similar to libmad, libmpcdec provides samples with higher quality than 16 bit. Send 24 bit samples to MPD, which allows MPD to apply dithering just in case the output devices are only 16 bit capable.
* mpc: moved sample size into a constantMax Kellermann2008-10-301-5/+9
| | | | Don't hard-code the "16 bits" or "2 bytes" in multiple locations.
* mpc: renamed variable "s16" to "dest"Max Kellermann2008-10-301-5/+4
| | | | | The name "s16" implies 16 bit integers. To make the code more generic, rename it to "dest".
* mpc: fix broken integer sample conversionMax Kellermann2008-10-301-6/+4
| | | | | | The conversion of integer samples was completely broken, which presumably didn't annoy anybody because libmpcdec provides float samples on most installations.
* mpc: fix boolean interpretation of input_stream_seek()Max Kellermann2008-10-301-1/+1
| | | | | When input_stream_seek() was converted to return a bool, this wasn't adjusted in the musepack plugin.
* decoder: use bool for return values and flagsMax Kellermann2008-10-3018-144/+148
| | | | | Don't return 0/-1 on success/error, but true/false. Instead of int, use bool for storing flags.
* mp3: make mp3_read() return boolMax Kellermann2008-10-301-19/+7
| | | | | | | Its only caller in mp3_decode() just compared its value with DECODE_BREAK. Convert that to bool, and return false if the loop should be ended. Also eliminate some superfluous command checking code, which was already done in the preceding while loop.
* output: removed audio_output.resultMax Kellermann2008-10-294-27/+12
| | | | | Since open() and play() close the device on error, we can simply check audio_output.open instead of audio_output.result after a call.
* output: delay reopen after device failureMax Kellermann2008-10-295-3/+24
| | | | | | | When one of several output devices failed, MPD tried to reopen it quite often, wasting a lot of resources. This patch adds a delay: wait 10 seconds before retrying. This might be changed to exponential delays later, but for now, it makes the problem go away.
* output: moved code to audio_output_update()Max Kellermann2008-10-293-8/+19
| | | | Moved code from syncAudioDeviceStates() to audio_output_update().
* output: always call cancel() before stop()Max Kellermann2008-10-293-14/+7
| | | | | | Stopping an audio output device without cancelling its buffer doesn't make sense. Combine the two operations, which saves several cancel calls.
* input_file: refuse to open non-regular filesMax Kellermann2008-10-291-0/+7
| | | | | Don't allow users to open a file which is non-regular (e.g. pipes, devices).
* input_file: check fstat() failureMax Kellermann2008-10-291-0/+6
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* input_file: use GLib for loggingMax Kellermann2008-10-291-3/+3
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* log: don't use utils.hMax Kellermann2008-10-293-27/+17
| | | | Prefer GLib over utils.h.
* output: moved audioDeviceStates to audio_output.enabledMax Kellermann2008-10-293-20/+12
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* output: close device on play errorMax Kellermann2008-10-297-12/+7
| | | | | When an output plugin fails to play a chunk, close it. This replaces various manual close() calls in nearly all plugins.
* output: use bool for return values and flagsMax Kellermann2008-10-2921-153/+162
| | | | | Don't return 0/-1 on success/error, but true/false. Instead of int, use bool for storing flags.
* audio: call closeAudioDevice() on errorMax Kellermann2008-10-291-5/+2
| | | | | Instead of implementing another loop which calls audio_output_close() on all configured devices, simply call closeAudioDevice().
* audio: removed variable "audioOpened"Max Kellermann2008-10-291-9/+1
| | | | This variable is never read, and can be removed.
* pulse: clear pa_simple pointer in pulse_close()Max Kellermann2008-10-291-0/+1
| | | | | The pointer becomes invalid due to pa_simple_free(), which may lead to segmentation faults when the output is reopened later.
* pulse: use GLib instead of util.h/log.hMax Kellermann2008-10-291-28/+25
| | | | Use GLib allocation and logging functions.
* pulse: no CamelCaseMax Kellermann2008-10-292-50/+51
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* player: handle songs shorter than the initial bufferMax Kellermann2008-10-291-1/+2
| | | | | | | When the decoder exited before the buffer has grown big enough ("buffer_before_play"), the player thread waited forever. Add an additional check which disables buffering as soon as the decoder exits.
* oss: check and override audio_format properlyMax Kellermann2008-10-291-0/+7
| | | | | Don't accept 24 bit audio. Force MPD to use 16 bit if anything other than 8 or 16 bit is selected.
* player: added variable "play_audio_format"Max Kellermann2008-10-291-2/+4
| | | | | | | | | The local variable "play_audio_format" is updated every time the player starts playing a new song. This way, we always know exactly which audio format is current. The old code broke when a new song had a different format: ob.audio_format is the format of the next song, not of the current one - using this caused breakage for the software volume control.
* decoder: automatically flush the output buffer after decoder exitsMax Kellermann2008-10-2912-30/+3
| | | | | | | A decoder_flush() invocation was missing in the FLAC plugin, resulting in casual assertion failures due to a wrong assumption about the last chunk's audio format. It's much easier to remove that decoder_flush() function and make the decoder thread call ob_flush().
* decoder_api: make the audio_format parameter mandatoryMax Kellermann2008-10-291-5/+3
| | | | | There are no callers which pass audio_format==NULL, and it shouldn't be allowed.
* player: request new song only if there is no pending commandMax Kellermann2008-10-291-1/+2
| | | | | | | Request the next song from the playlist (by clearing pc.next_song) only if the player command is empty. If it is not, the player may be clearing the song that has already been queued, leading to an assertion failure.
* decoder_api: removed decoder_clear()Max Kellermann2008-10-2910-23/+7
| | | | | Call ob_clear() in decoder_command_finished() instead of implementing that call in every decoder plugin.
* mp3: seek in backgroundMax Kellermann2008-10-291-5/+7
| | | | | | | | | | Remember the seek_where argument and call decoder_command_finished() immediately. This way, the player thread can continue working, and we can receive more commands. This also fixes several issues which resulted in broken frames, leading to erroneos "elapsed" values: frames weren't parsed properly, since the code was checking for command!=NONE.
* mp3: return from mp3_synth_and_send() on any commandMax Kellermann2008-10-291-2/+2
| | | | | Previously, the function would only return when a STOP was issued. It makes more sense to consider all possible commands.
* mp3: moved code to mp3_synth_and_send()Max Kellermann2008-10-291-60/+74
| | | | Break the large function mp3_read() into smaller pieces.
* mp3: moved code to mp3_send_pcm()Max Kellermann2008-10-291-29/+48
| | | | Break the large function mp3_read() into smaller pieces.
* mp3: moved code to mp3_update_timer_next_frame()Max Kellermann2008-10-291-16/+28
| | | | Break the large function mp3_read() into smaller pieces.
* mp3: moved code to mp3_this_frame_offset()Max Kellermann2008-10-291-10/+10
| | | | Break the large function mp3_read() into smaller pieces.
* mp3: moved code to mp3_time_to_frame()Max Kellermann2008-10-291-11/+21
| | | | Break the large function mp3_read() into smaller pieces.
* mp3: assert that the stream is seekableMax Kellermann2008-10-291-5/+5
| | | | | | dc_seek() won't send a SEEK command to the decoder thread unless the stream is seekable. No need to do another check; convert that to an assertion.
* decoder_api: don't cast need_chunks() to intMax Kellermann2008-10-291-4/+4
| | | | | need_chunks() returns a decoder_command enum. Store its return value as this type.
* mp3: moved code to mp3_filesize_to_song_length()Max Kellermann2008-10-281-23/+42
| | | | | The function mp3_decode_first_frame() is too large. Move some code to separate smaller functions.
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