Merge branch 'v0.21.x'

NEWS

@@ -9,6 +9,8 @@ ver 0.22 (not yet released)
   - hdcd: new plugin based on FFmpeg's "af_hdcd" for HDCD playback
 
 ver 0.21.9 (not yet released)
+* input
+  - buffer: fix deadlock bug
 * Android
   - fix crash on ARMv7
   - request storage permission on Android 6+

doc/mpd.conf.5

@@ -140,7 +140,6 @@ of database.
 .B auto_update_depth <N>
 Limit the depth of the directories being watched, 0 means only watch
 the music directory itself.  There is no limit by default.
-.TP
 .SH REQUIRED AUDIO OUTPUT PARAMETERS
 .TP
 .B type <type>
@@ -164,57 +163,12 @@ Specifies how replay gain is applied.  The default is "software",
 which uses an internal software volume control.  "mixer" uses the
 configured (hardware) mixer control.  "none" disables replay gain on
 this audio output.
-.SH OPTIONAL ALSA OUTPUT PARAMETERS
-.TP
-.B device <dev>
-This specifies the device to use for audio output.  The default is "default".
 .TP
 .B mixer_type <hardware, software or none>
 Specifies which mixer should be used for this audio output: the
 hardware mixer (available for ALSA, OSS and PulseAudio), the software
 mixer or no mixer ("none").  By default, the hardware mixer is used
 for devices which support it, and none for the others.
-.TP
-.B mixer_device <mixer dev>
-This specifies which mixer to use.  The default is "default".  To use
-the second sound card in a system, use "hw:1".
-.TP
-.B mixer_control <mixer ctrl>
-This specifies which mixer control to use (sometimes referred to as
-the "device").  The default is "PCM".  Use "amixer scontrols" to see
-the list of possible controls.
-.TP
-.B mixer_index <mixer index>
-A number identifying the index of the named mixer control.  This is
-probably only useful if your alsa device has more than one
-identically\-named mixer control.  The default is "0".  Use "amixer
-scontrols" to see the list of controls with their indexes.
-.TP
-.B auto_resample <yes or no>
-Setting this to "no" disables ALSA's software resampling, if the
-hardware does not support a specific sample rate.  This lets MPD do
-the resampling.  "yes" is the default and allows ALSA to resample.
-.TP
-.B auto_channels <yes or no>
-Setting this to "no" disables ALSA's channel conversion, if the
-hardware does not support a specific number of channels.  Default: "yes".
-.TP
-.B auto_format <yes or no>
-Setting this to "no" disables ALSA's sample format conversion, if the
-hardware does not support a specific sample format.  Default: "yes".
-.TP
-.B buffer_time <time in microseconds>
-This sets the length of the hardware sample buffer in microseconds.  Increasing
-it may help to reduce or eliminate skipping on certain setups.  Most users do
-not need to change this.  The default is 500000 microseconds (0.5 seconds).
-.TP
-.B period_time <time in microseconds>
-This sets the time between hardware sample transfers in microseconds.
-Increasing this can reduce CPU usage while lowering it can reduce underrun
-errors on bandwidth-limited devices.  Some users have reported good results
-with this set to 50000, but not all devices support values this high.  Most
-users do not need to change this.  The default is 256000000 / sample_rate(kHz),
-or 5804 microseconds for CD-quality audio.
 .SH FILES
 .TP
 .BI ~/.mpdconf

doc/protocol.rst

@@ -178,8 +178,9 @@ of:
   file's time stamp with the given value (ISO 8601 or UNIX
   time stamp).
 
-- ``(AudioFormat == 'SAMPLERATE:BITS:CHANNELS')``:
+- ``(AudioFormat == 'SAMPLERATE:BITS:CHANNELS')``: compares the audio
-  compares the audio format with the given value.
+  format with the given value.  See :ref:`audio_output_format` for a
+  detailed explanation.
 
 - ``(AudioFormat =~ 'SAMPLERATE:BITS:CHANNELS')``:
   matches the audio format with the given mask (i.e. one
@@ -423,7 +424,9 @@ Querying :program:`MPD`'s status
     - ``xfade``: ``crossfade`` in seconds
     - ``mixrampdb``: ``mixramp`` threshold in dB
     - ``mixrampdelay``: ``mixrampdelay`` in seconds
-    - ``audio``: The format emitted by the decoder plugin during playback, format: ``*samplerate:bits:channels*``. Check the user manual for a detailed explanation.
+    - ``audio``: The format emitted by the decoder plugin during
+      playback, format: ``samplerate:bits:channels``.  See
+      :ref:`audio_output_format` for a detailed explanation.
     - ``updating_db``: ``job id``
     - ``error``: if there is an error, returns message here
 

doc/user.rst

@@ -402,14 +402,9 @@ The following table lists the audio_output options valid for all plugins:
      - The name of the plugin
    * - **name**
      - The name of the audio output. It is visible to the client. Some plugins also use it internally, e.g. as a name registered in the PULSE server.
-   * - **format**
+   * - **format samplerate:bits:channels**
-     -  Always open the audio output with the specified audio format samplerate:bits:channels), regardless of the format of the input file. This is optional for most plugins.
+     -  Always open the audio output with the specified audio format, regardless of the format of the input file. This is optional for most plugins.
-
+        See :ref:`audio_output_format` for a detailed description of the value.
-        Any of the three attributes may be an asterisk to specify that this attribute should not be enforced, example: 48000:16:*. *:*:* is equal to not having a format specification.
-
-        The following values are valid for bits: 8 (signed 8 bit integer samples), 16, 24 (signed 24 bit integer samples padded to 32 bit), 32 (signed 32 bit integer samples), f (32 bit floating point, -1.0 to 1.0), "dsd" means DSD (Direct Stream Digital). For DSD, there are special cases such as "dsd64", which allows you to omit the sample rate (e.g. dsd512:2 for stereo DSD512, i.e. 22.5792 MHz).
-
-        The sample rate is special for DSD: :program:`MPD` counts the number of bytes, not bits. Thus, a DSD "bit" rate of 22.5792 MHz (DSD512) is 2822400 from :program:`MPD`'s point of view (44100*512/8).
    * - **enabed yes|no**
      - Specifies whether this audio output is enabled when :program:`MPD` is started. By default, all audio outputs are enabled. This is just the default setting when there is no state file; with a state file, the previous state is restored.
    * - **tags yes|no**
@@ -504,10 +499,31 @@ reference.
 Audio Format Settings
 ---------------------
 
+.. _audio_output_format:
+
 Global Audio Format
 ~~~~~~~~~~~~~~~~~~~
 
-The setting audio_output_format forces :program:`MPD` to use one audio format for all outputs. Doing that is usually not a good idea. The values are the same as in format in the audio_output section.
+The setting ``audio_output_format`` forces :program:`MPD` to use one
+audio format for all outputs.  Doing that is usually not a good idea.
+
+The value is specified as ``samplerate:bits:channels``.
+
+Any of the three attributes may be an asterisk to specify that this
+attribute should not be enforced, example: ``48000:16:*``.
+``*:*:*`` is equal to not having a format specification.
+
+The following values are valid for bits: ``8`` (signed 8 bit integer
+samples), ``16``, ``24`` (signed 24 bit integer samples padded to 32
+bit), ``32`` (signed 32 bit integer samples), ``f`` (32 bit floating
+point, -1.0 to 1.0), ``dsd`` means DSD (Direct Stream Digital). For
+DSD, there are special cases such as ``dsd64``, which allows you to
+omit the sample rate (e.g. ``dsd512:2`` for stereo DSD512,
+i.e. 22.5792 MHz).
+
+The sample rate is special for DSD: :program:`MPD` counts the number
+of bytes, not bits. Thus, a DSD "bit" rate of 22.5792 MHz (DSD512) is
+2822400 from :program:`MPD`'s point of view (44100*512/8).
 
 Resampler
 ~~~~~~~~~
@@ -885,7 +901,7 @@ To verify if :program:`MPD` converts the audio format, enable verbose logging, a
 .. code-block:: none
 
     decoder: audio_format=44100:24:2, seekable=true
-    output: opened plugin=alsa name="An ALSA output"audio_format=44100:16:2
+    output: opened plugin=alsa name="An ALSA output" audio_format=44100:16:2
     output: converting from 44100:24:2
 
 This example shows that a 24 bit file is being played, but the sound chip cannot play 24 bit. It falls back to 16 bit, discarding 8 bit.
@@ -912,7 +928,7 @@ Check list for bit-perfect playback:
   device (:samp:`hw:0,0` or similar).
 * Don't use software volume (setting :code:`mixer_type`).
 * Don't force :program:`MPD` to use a specific audio format (settings
-  :code:`format`, :code:`audio_output_format`).
+  :code:`format`, :ref:`audio_output_format <audio_output_format>`).
 * Verify that you are really doing bit-perfect playback using :program:`MPD`'s verbose log and :file:`/proc/asound/card*/pcm*p/sub*/hw_params`. Some DACs can also indicate the audio format.
 
 Direct Stream Digital (DSD)

src/input/BufferedInputStream.cxx

@@ -165,6 +165,30 @@ BufferedInputStream::RunThread() noexcept
 			idle = false;
 			seek = false;
 			client_cond.notify_one();
+		} else if (!idle && !read_error &&
+			   offset != input->GetOffset() &&
+			   !IsAvailable()) {
+			/* a past Seek() call was a no-op because data
+			   was already available at that position, but
+			   now we've reached a new position where
+			   there is no more data in the buffer, and
+			   our input is reading somewhere else (maybe
+			   stuck at the end of the file); to find a
+			   way out, we now seek our input to our
+			   reading position to be able to fill our
+			   buffer */
+
+			try {
+				input->Seek(lock, offset);
+			} catch (...) {
+				/* this is really a seek error, but we
+				   register it as a read_error,
+				   because seek_error is only checked
+				   by Seek(), and at our frontend (our
+				   own InputStream interface) is in
+				   "read" mode */
+				read_error = std::current_exception();
+			}
 		} else if (!idle && !read_error &&
 			   input->IsAvailable() && !input->IsEOF()) {
 			const auto read_offset = input->GetOffset();