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* curl: read stream name from HTTP response headersMax Kellermann2008-10-261-0/+5
| | | | | If the server sends the headers "icy-name", "ice-name", "x-audiocast-name", set the stream title.
* input_curl: set input_stream.mime from Content-Type response headerMax Kellermann2008-10-261-0/+3
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* input_curl: added local variable "value"Max Kellermann2008-10-261-14/+22
| | | | | | Replaced the local variable "colon" (which had only temporary meaning) with the variable "value". It is a pointer to the first byte of the header value.
* input_stream: use g_free() in input_stream_close()Max Kellermann2008-10-261-8/+5
| | | | g_free() allows passing the NULL pointer.
* input_stream: use "bool" instead of "int"Max Kellermann2008-10-2614-49/+55
| | | | | For boolean values and success flags, use bool instead of integer (1/0 for true/false, 0/-1 for success/failure).
* input_stream: input_stream_close() returns voidMax Kellermann2008-10-264-13/+7
| | | | close() shouldn't fail with read-only streams.
* input_stream: added struct input_pluginMax Kellermann2008-10-266-56/+63
| | | | | | | Instead of managing a set of method pointers in each input_stream struct, move these into the new input_plugin struct. Each input_stream has only a pointer to the plugin struct. Pointers to all implementations are kept in the array "input_plugins".
* input_stream: no CamelCaseMax Kellermann2008-10-2617-160/+158
| | | | Renamed all functions and variables.
* input_file: removed global constructorMax Kellermann2008-10-263-7/+0
| | | | The global constructor is empty, and can be removed.
* input_file: don't export internal methodsMax Kellermann2008-10-262-19/+25
| | | | | The methods are only used in inputStream_fileOpen(), and should not be exported.
* input_stream: removed the InputStream typedefMax Kellermann2008-10-2613-50/+65
| | | | Everybody should use struct input_stream.
* input_stream: renamed sources, no CamelCaseMax Kellermann2008-10-268-11/+11
| | | | Renamed inputStream.c and inputStream_file.c.
* http: use libcurlMax Kellermann2008-10-267-1171/+525
| | | | | | | MPD's HTTP client code has always been broken, no matter how effort was put into fixing it. Replace it with libcurl, which is known to be quite stable. This adds a fat library dependency, but only for people who need streaming.
* input_stream: added input_stream_global_finish()Max Kellermann2008-10-263-0/+7
| | | | | The hook input_stream_global_finish() deinitializes global structures of all input stream implementations.
* stored_playlist: fixed signed comparison warningMax Kellermann2008-10-261-1/+1
| | | | Cast playlist_max_length to off_t before comparing it to stat.st_size.
* command: removed range check from check_bool()Max Kellermann2008-10-261-6/+0
| | | | check_bool() accepts only "0" or "1". The range check is superfluous.
* renamed src/audioOutputs/ to src/output/Max Kellermann2008-10-2614-19/+18
| | | | Again, no CamelCase in the directory name.
* renamed src/inputPlugins/ to src/decoder/Max Kellermann2008-10-2616-14/+14
| | | | | These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name.
* removed internal copy of libmp4ffMax Kellermann2008-10-2515-4645/+2
| | | | | | MPD shouldn't integrate sources of other libraries. Since libmp4ff is part of libfaad, we should remove the old copy from src/mp4ff and link with the current version from libfaad instead.
* pulse: force 16 bit audio sample formatMax Kellermann2008-10-251-5/+3
| | | | | | | | | PA_SAMPLE_S16NE is the only sample format which is suported by both MPD and pulseaudio. Unfortunately, pulse does not accept 24 bit samples. Instead of bailing out with an error message, we should tell the MPD core to convert all samples to 16 bit for pulse.
* player: don't clear command before do_play() returnsMax Kellermann2008-10-241-1/+0
| | | | | | | | | This bug caused the audio output devices to stay open, although MPD wasn't playing: quitDecode() resetted player_control.command, assuming that the command was STOP. This way, player_task() didn't see the CLOSE_AUDIO command, and the device was kept open. Don't clear player_control.command in quitDecode().
* remove unused sourcesMax Kellermann2008-10-242-221/+0
| | | | These are results from failed merges which I didn't notice.
* jack: support for 24 bit samplesMax Kellermann2008-10-241-1/+32
| | | | | | When the audio source provides 24 bit samples, don't bother to convert (lossily) them to 16 bit before jack's floating point conversion - go directly from 24 bit to float.
* jack: moved code to jack_write_samples_16()Max Kellermann2008-10-241-14/+42
| | | | Move sample format dependent code to a separate function.
* jack: eliminated CamelCaseMax Kellermann2008-10-241-73/+76
| | | | | Renamed all variables and functions. Add the prefix "mpd_jack_" to function names.
* jack: added assertions against partial framesMax Kellermann2008-10-241-0/+2
| | | | We must never pass partial frames. Added assertions to debug this.
* jack: optimize local variablesMax Kellermann2008-10-241-9/+8
| | | | | Merge the variables "avail_data" and "avail_frames" into "available". Both variables are never used at the same time.
* jack: lockless data transfer to jack threadMax Kellermann2008-10-241-47/+15
| | | | | | | | The JACK documentation postulates that the process() callback must not block, therefore locking is forbidden. Anyway, the old code was racy. Remove all locks, and don't wait for more data to become available - just send to the port what is already in the buffer.
* jack: partial writes to ring bufferMax Kellermann2008-10-241-15/+15
| | | | | | Don't wait until there is room for the full data chunk passed to jack_playAudio(). Try to incrementally send as much as possible into the ring buffer.
* jack: added constant "frame_size"Max Kellermann2008-10-241-1/+2
| | | | | | Don't hard-code a frame size of "4" (16 bit stereo), calculate the sample size from sizeof(*buffer), and create the constant "frame_size".
* jack: fix indentationMax Kellermann2008-10-241-33/+33
| | | | Indent with tabs.
* pcm_resample: support for libsamplerate < 0.1.3Max Kellermann2008-10-241-0/+20
| | | | | libsamplerate 0.1.2 didn't have the 32 bit <-> float conversion routines. Emulate them in case they aren't supported.
* player: don't send partial frames of silenceMax Kellermann2008-10-231-1/+7
| | | | | | Another partial frame fix: the silence buffer was 1020 bytes, which had room for 127.5 24 bit stereo frames. Don't send the partial last frame in this case.
* pcm_utils: added 24 bit conversion functionsMax Kellermann2008-10-231-0/+108
| | | | | | 24 bit output is as important as 16 bit output. Provide a pcm_convert() implementation which can convert to 24 bit with as little quality loss as possible.
* pcm_utils: generic pcm_convert_size() implementationMax Kellermann2008-10-231-21/+2
| | | | | | The old pcm_convert_size() ignored most of the destination format, e.g. it did not check its sample size, and assumed it is 16 bit. Simplify and universalize it by using audio_format_frame_size().
* pcm_utils: moved code to pcm_convert_16()Max Kellermann2008-10-231-21/+39
| | | | | pcm_convert() converted only to 16 bit. To be able to support other sample sizes, move that 16 bit specific code to a separate function.
* pcm_channels: added 24 bit implementationsMax Kellermann2008-10-232-0/+84
| | | | | The 24 bit implementation is mostly copy'n'paste of the 16 bit version, except that the data type is int32_t instead of int16_t.
* pcm_utils: moved channel conversion functions to pcm_channels.cMax Kellermann2008-10-234-82/+138
| | | | Separate code from pcm_utils.c to keep it small and simple.
* pcm_resample: implemented 24 bit resamplingMax Kellermann2008-10-233-0/+94
| | | | | | | Similar to pcm_resample_16(), implement pcm_resample_24(). The 24 bit implementation is very similar, but it uses src_int_to_float_array() instead of src_short_to_float_array() before sending data to libsamplerate.
* pcm_resample: moved code to pcm_resample_set()Max Kellermann2008-10-231-29/+40
| | | | | A future patch will implement a 24 bit resampler. To unify code, move code which can be shared to a separate function.
* pcm_resample: eliminated "sample" local variablesMax Kellermann2008-10-231-9/+3
| | | | | Copy from source to destination buffer directly, don't use the temporary variables "lsample" and "rsample".
* pcm_resample: don't resample partial samplesMax Kellermann2008-10-232-0/+9
| | | | | Added assertions which ensure that there are no partial samples in the source buffer.
* pcm_resample: don't hard-code sample sizeMax Kellermann2008-10-232-4/+4
| | | | | | Use sizeof(sample) instead of hard-coding "2". Although we're in 16 bit right now, this will make code sharing easier when we support other sample sizes.
* pcm_utils: moved code to pcm_resample.cMax Kellermann2008-10-237-176/+306
| | | | | | Separate the resampling code from the rest of pcm_utils.c. Create two sub-libraries: pcm_resample_libsamplerate.c and pcm_resample_fallback.c.
* command: fix boolean value parserMax Kellermann2008-10-231-1/+1
| | | | | Due to a logic error, no value was valid for the boolean value parser. Replace "||" with "&&".
* mp3: send 24 bit PCM dataMax Kellermann2008-10-231-63/+24
| | | | | | | libmad produces samples of more than 24 bit. Rounding that down to 16 bits using dithering makes those people lose quality who have a 24 bit capable sound device. Send 24 bit PCM data, and let the receiver decide whether to apply 16 bit dithering.
* mp3: use sizeof(sample) instead of hard-coded "2"Max Kellermann2008-10-231-2/+3
| | | | | We are going to convert the code to 24 bit; don't hard-code a sample size of 2 bytes.
* pcm_dither: added generic 24 to 16 bit ditheringMax Kellermann2008-10-235-8/+139
| | | | | Copied and adapted code from the mp3 decoder plugin. This library now replaces the old and low-quality function pcm_convert_24_to_16().
* audio: allow 24 and 8 bit outputMax Kellermann2008-10-231-4/+2
| | | | | | | I added 24 bit support a while ago, but it wasn't possible to force 24 bit output. Add 24 and 8 bit to the list of allowed sample sizes. Although 8 bit audio isn't as widely used as 24 bit, there is no reason to exclude it.
* output_buffer: don't split framesMax Kellermann2008-10-233-1/+16
| | | | | | | Splitting a frame between two buffer chunks causes distortion in the output. MPD used to assume that the chunk size 1020 would never cause splitted frames, but that isn't the case for 24 bit stereo (127.5 frames), and even less for files with even more channels.
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