Perex at 06:54, 4 April 2018

2018-04-04T06:54:00Z

New page

{| align="right"
| __TOC__
|}
=Detailed changelog between 1.0.28 and 1.1.6 releases=
=Changelog between 1.0.29 and 1.1.6 releases=
==alsa-oss==
===Core===
: - Release v1.1.6
:
: - Change FSF address (Franklin Street)
:
===Documentation===
: - Change FSF address (Franklin Street)
:
===Mixer Emulation===
: - Change FSF address (Franklin Street)
:
===OSS Redirection===
: - Change FSF address (Franklin Street)
:
===PCM Emulation===
: - Change FSF address (Franklin Street)
:
=Detailed changelog between 1.1.5 and 1.1.6 releases=
=Changelog between 1.0.29 and 1.1.6 releases=
==alsa-lib==
===Core===
: - Release v1.1.6
:
: - a set of fixes to reduce gcc warnings
:
: - modules: smixer_python - add support for python3
:
: - conf/ucm: Add chtrt5645-mono-speaker-analog-mic configuration
:
: Add a configuration for Cherry Trail boards which use a rt5645 codec
: connected to a mono speaker and with an analog mic on IN2N + IN2P.
:
: The chtrt5645-mono-speaker-analog-mic/HiFi.conf for this is based on the
: latest version from https://github.com/plbossart/UCM/tree/master/chtrt5645
: with all the unused input options removed and some changes made to make
: the analog mic work.
:
: This commit also adds 2 ucm dirs with the longname of 2 boards known to use
: this setup, which simply contain a symlink to the generic
: chtrt5645-mono-speaker-analog-mic entry.
:
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
: - conf/ucm: Add Gigabyte mobo UCM profile with dual HD-audio codecs
:
: Like the previous Lenovo laptops, some Gigabyte mobos have dual
: HD-audio codecs and need to switch dynamically via UCM profile.
: Reuse the same profile as Lenovo dual codecs, so far.
:
: - conf/ucm: Add dual HD-audio codecs config for Lenovo
:
: Some recent Lenovo laptops have dual codecs and we need to switch them
: accordingly. The kernel side already contains a fix and gives the
: unique longname string for identifying the board, and here we hook up
: the corresponding UCM profile.
:
: The profile was corrected and tested by Hui Wang on Lenovo p520.
:
: Tested-by: Kailang <kailang@realtek.com>
: Tested-by: Hui Wang <hui.wang@canonical.com>
: - conf/ucm: Rearrange Makefile.am
:
: Just rearrange and split to multiple lines to make future patches
: cleaner. No actual code change.
:
: - configure.ac: do not enable alisp code by default
:
: - configure.ac: build extra mixer modules conditionally
:
: - Change snd_dlopen() function to return the error string
:
: The dlopen() function might fail also for another reason than
: a missing file, thus return the error string from dlerror().
:
: - Change FSF address (Franklin Street)
:
===Control API===
: - control ext: fix the default .rawmidi_next_device callback
:
: The previous default settings caused an endless loop.
:
: - control: Proper reference of internal versioned functions
:
: The multiply defined versioned symbols have to be called with
: INTERNAL() wrapper.
:
: Add the missing declarations of the internal forms of
: snd_ctl_elem_info_get_dimension*() in the local header, and use them
: in the (still remaining) callers in alisp.
:
===HWDEP API===
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
===Mixer API===
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
===PCM API===
: - pcm: ioplug: Implement proper drain behavior
:
: This patch fixes the draining behavior of ioplug in the following
: ways:
:
: - When no draining ioplug callback is defined, implement the draining
: loop using snd_pcm_wait*() and sync with the drain finishes.
: This is equivalent with the implementation in the kernel write().
: Similarly as in kernel code, for non-blocking mode, it returns
: immediately after setting DRAINING state.
:
: - The hw_ptr update function checks the PCM state and stops the stream
: if the draining finishes.
:
: - When draining ioplug callback is defined, leave the whole draining
: operation to it. The callback is supposed to return -EAGAIN for
: non-blocking case, too.
:
: - When an error happens during draining, it drops the stream, for a
: safety reason.
:
: - pcm: Skip avail_min check during draining
:
: snd_pcm_wait() & co checks the current avail value and returns
: immediately if it satisfies <= avail_min condition. It's good in
: general except for one situation: draining. When the draining is
: being performed in the non-blocking mode, apps are supposed to wait
: via poll(), typically via snd_pcm_wait(). So this ends up with the
: busy loop because of the immediate return from snd_pcm_wait().
:
: A simple workaround is to put the PCM state check and ignore the
: avail_min condition if it's DRAINING state. The equivalent check is
: found in the kernel xfer code, too.
:
: - a set of fixes to reduce gcc warnings
:
: - pcm: hw: Keep control data from kernel when SND_PCM_APPEND
:
: Without this fix the application pointer would be reseted
: whenever an application opens a device with SND_PCM_APPEND.
:
: This would result in an Xrun if the device is already opened and
: in running state and the appl_ptr is use.
:
: - pcm: ioplug: update prepare and draining state correctly
:
: PREPARED should only be set when it is done and it was successfully.
:
: DRAINING should be signalled when starting to drain. There is no need to
: check if draining was successfully because it will change to drop (SETUP)
: in any case.
:
: - pcm: Provide areas_copy function which handles buffer wrap around
:
: The already existing areas_copy functions do not care about the end of
: the source and destination buffer.
: Therefore the caller has to take care that the requested offset+size
: is not exceeding any buffer limit.
:
: This additional function will take care about the end of an buffer
: and will continue at the beginning of the buffer.
: For example this is required when copying between buffers with
: different sizes (not multiple of).
: This is often the case in IO plugins like the JACK plugin.
:
: - pcm: ioplug: Provide hw_avail helper function for plugins
:
: This function can be called without calling snd_pcm_avail_update().
:
: The call to snd_pcm_avail_update() can take some time.
: Therefore some developers would not like to call it from a real-time
: context (e.g. from JACK client context).
:
: - pcm: Do not access lock_enabled if thread safe API
:
: Without this commit compiling fails when THREAD_SAFE_API is not
: enabled.
:
: - pcm: ioplug: Use boundary for wrap around
:
: if requested by the IO plugin
:
: Without this changes an IO plugin is not able to report
: that buffer_size frames were read from the buffer.
: When the buffer was full this is a valid action and
: has not to be handled as an under run.
:
: For example when the hw_ptr will be updated with
: hw_ptr += buffer_size
: and it is using the buffer_size as wrap around
: hw_ptr %= buffer_size
: would result in the same value as before the add operation.
:
: - pcm: another fix for the snd_pcm_area_silence() fast path
:
: The 64-bit fast path can be used only in limited conditions:
:
: - destination must be aligned to 64-bit (CPU aligned access)
: - step must be equal to width
: - physical with must be different than 24 (cannot be multiplied to 64)
:
: - pcm: fix a bug to copy silent samples aligned to 64
:
: bits for
: 24 bit sample cases
:
: A function of 'snd_pcm_area_silence()' has a fast path to copy silent data
: efficiently. However, the fast path works well just for a case that target
: buffer consists of data samples for which unit of data alignment is
: divisors of 64 bits.
:
: At present, the fast path handles sample data aligned to 24 bit. In this
: case, the buffer can includes extra 8 bits. This has no issue for 'signed'
: case because silent data is zero, however it has an issue for 'unsigned'
: case.
:
: This commit fixes the bug by skipping cases of sample data of 24 bit.
:
: - pcm: Fix two bugs in snd_pcm_area_silence()
:
: First, after silencing the buffer 64 bits at a time, any remaining samples
: need to be silenced by the following width-specific code. However, instead
: of silencing the end of the buffer, the code instead re-silences the start
: of the buffer, leaving the end unsilenced. To fix this, update the pointer
: used by the width-specific code to point to the end of the area just
: silenced, instead of leaving it pointing to the start of the buffer.
:
: Second, the code for 24 bit samples can only silence a single sample, as
: there's no loop for multiple samples as with other formats. To fix this,
: add a loop similar to the ones used for every other width.
:
: The symptoms of these bugs are random data at the end of every supposedly
: silenced buffer with certain format/buffer size combinations, resulting in
: pops and noise.
:
: Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
: - pcm: Return the consistent error code for unexpected PCM states
:
: Some PCM functions have the sanity check of the expected PCM states,
: and most of them return -EBADFD if the current state doesn't match.
: This is bad for some programs like aplay that expect the function
: returning a proper code corresponding to the state, e.g. -ESTRPIPE for
: the suspend.
:
: This patch is an attempt to address such inconsistencies. The sanity
: checker bad_pcm_state() now returns the error code instead of bool, so
: that the caller can pass the returned code as is. And it calls a new
: helper, pcm_state_to_error(), for obtaining the error code to certain
: known PCM error state.
:
: While we're at it, use the new pcm_state_to_error() for simplifying
: the existing code to retrieve the error code, too.
:
: Tested-by: Mirza Krak <mirza.krak@gmail.com>
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
: - timer: Proper reference of internal versioned symbols
:
: The multiply defined versioned symbols have to be called with
: INTERNAL() wrapper.
:
: Add the missing declarations of versioned timer API functions in the
: local header, and use them in the callers in PCM.
:
: - pcm: fix wrong comments for some cases of linear interpolation of PCM samples
:
: A commit 16b3bf447c28 ('Enhanced bitmasks in PCM - added support for more
: formats by Takashi and me') adds support for some cases of linear
: interpolation of PCM samples, however some of added comments are not
: proper. This commit fixes them.
:
: - pcm: remove unused macros of NORMS_LABELS/NORMS_END
:
: A commit fcd164e6229c ("Permit to PCM plug configuration to specify unchanged
: parameters. Added support for RT signals to async interface. Added ops for
: PCM mix.") added a pair of NORMS_LABELS/NORMS_END, however they have been
: no longer used.
:
: This commit removes them in a point to reduce the amount of code to
: maintain.
:
: - pcm: remove unused macros of GETU_LABELS/GETU_END
:
: A commit 07c07da44f27 ("Fixed signess for route conversion") obsoletes
: usage of a pair of GETU_LABEL/GETU_END, but it did not remove some
: actual codes in 'src/pcm_plugin_ops.h'.
:
: This commit removes them in a point to reduce the amount of code to
: maintain.
:
: - pcm: remove unused macros of COPY_LABELS/COPY_END
:
: A commit 7b054f4dce56 obsoleted usage of a pair of COPY_LABELS/COPY_END,
: however it did not remove some codes in 'src/pcm/plugin_ops.h'.
:
: This commit removes them in a point to reduce the amount of code to
: maintain.
:
: - pcm: plug: add SND_PCM_FORMAT_{S, U}20 to linear_preferred_formats
:
: This commit adds the recently added formats SND_PCM_FORMAT_{S,U}20 to
: the linear_preferred_formats array in pcm_plug.
: Let's give them lower priority than more standard S24 formats but a higher
: priority than less typical 3-byte versions.
:
: Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
: - pcm: linear, route: handle linear formats with 20-bit sample on 4 bytes
:
: The previous patch has added 20-bit PCM formats SND_PCM_FORMAT_{S,U}20 to
: alsa-lib.
: We need to extend the linear format conversion code with handling of these
: sample formats so they can also be utilized by applications that only
: recognize the more typical ones like SND_PCM_FORMAT_S16.
:
: Since the conversion arrays are indexed by a format bit width divided by 8
: the easiest way to handle these formats is to treat them like they were
: 40-bit wide (the next free integer multiple of 8 bits).
: This doesn't create a collision risk with a future format since there can't
: really be a 40-bit sample format that occupies 4 bytes.
:
: Make sure we use the getput conversion method for these formats since a
: direct conversion from / to them is not supported.
:
: Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
: - pcm: add and describe SND_PCM_FORMAT_{S, U}20
:
: This patch adds and describes in various functions that query format
: properties SND_PCM_FORMAT_{S,U}20 formats that were recently added to the
: kernel as SNDRV_PCM_FORMAT_{S,U}20.
:
: These formats are similar to existing 20-bit PCM formats
: SND_PCM_FORMAT_{S,U}20_3, however they occupy 4 bytes instead of 3.
:
: Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
: - pcm: route: Fix use_getput flag computation for 3 byte formats
:
: Commit de63b942acf520 ("pcm: route: Use get/put labels for all 3 byte formats")
: wanted to make the route plugin use get / put labels not only for 24-bit
: physical width formats but also for 18 and 20-bit ones.
: There was, however, a typo in that commit so a check for these widths
: didn't really work.
:
: Let's fix it now.
:
===Timer API===
: - timer: Proper reference of internal versioned symbols
:
: The multiply defined versioned symbols have to be called with
: INTERNAL() wrapper.
:
: Add the missing declarations of versioned timer API functions in the
: local header, and use them in the callers in PCM.
:
===Topology API===
: - a set of fixes to reduce gcc warnings
:
: - topology: Fix parsing config with multiple hw_configs
:
: Currently, if the config file includes several hw_configs sections,
: parse_hw_config_refs() returns after parsing only the first section.
:
: For example, the following config, based on
: alsa-lib/src/conf/topology/broadwell/broadwell.conf, is parsed incorrectly:
:
: ~~~~
: SectionHWConfig."CodecHWConfig" {
: id "1"
: format "I2S" # physical audio format.
: bclk "master" # Platform is master of bit clock
: fsync "master" # platform is master of fsync
: }
:
: SectionHWConfig."CodecHWConfig2" {
: id "2"
: format "AC97"
: }
:
: SectionLink."Codec" {
:
: # used for binding to the physical link
: id "0"
:
: hw_configs [
: "CodecHWConfig"
: "CodecHWConfig2"
: ]
:
: default_hw_conf_id "2"
: }
: ~~~~
:
: - topology: Fix to skip writing of header for compound elements
:
: While calculating the size of data to be written into the topology
: binary file, the size of the compound elements is added as well. This
: results in wrong file offset calculation and topology build failure.
:
: The compound elements shouldn't be written to the topology as these are
: already embedded as part of other elements. So, skip adding the size of
: compound elements to the file offset size calculation.
:
===ALSA Lisp===
: - control: Proper reference of internal versioned functions
:
: The multiply defined versioned symbols have to be called with
: INTERNAL() wrapper.
:
: Add the missing declarations of the internal forms of
: snd_ctl_elem_info_get_dimension*() in the local header, and use them
: in the (still remaining) callers in alisp.
:
: - Change FSF address (Franklin Street)
:
===ALSA Server===
: - a set of fixes to reduce gcc warnings
:
: - Change FSF address (Franklin Street)
:
===Compatibility routines===
: - Change FSF address (Franklin Street)
:
===Configuration===
: - conf: USB-Audio: Add second S/PDIF device on Phiree U2SX
:
: Phiree U2SX is the successor of Phiree U2 and has the same unusual
: configuration.
:
: See ea865bba4615d906144ae5d4f72a4aad2baffe1f for reference.
:
: - conf/ucm: Add chtrt5645-mono-speaker-analog-mic configuration
:
: Add a configuration for Cherry Trail boards which use a rt5645 codec
: connected to a mono speaker and with an analog mic on IN2N + IN2P.
:
: The chtrt5645-mono-speaker-analog-mic/HiFi.conf for this is based on the
: latest version from https://github.com/plbossart/UCM/tree/master/chtrt5645
: with all the unused input options removed and some changes made to make
: the analog mic work.
:
: This commit also adds 2 ucm dirs with the longname of 2 boards known to use
: this setup, which simply contain a symlink to the generic
: chtrt5645-mono-speaker-analog-mic entry.
:
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
: - conf/ucm: Add Gigabyte mobo UCM profile with dual HD-audio codecs
:
: Like the previous Lenovo laptops, some Gigabyte mobos have dual
: HD-audio codecs and need to switch dynamically via UCM profile.
: Reuse the same profile as Lenovo dual codecs, so far.
:
: - conf/ucm: Add dual HD-audio codecs config for Lenovo
:
: Some recent Lenovo laptops have dual codecs and we need to switch them
: accordingly. The kernel side already contains a fix and gives the
: unique longname string for identifying the board, and here we hook up
: the corresponding UCM profile.
:
: The profile was corrected and tested by Hui Wang on Lenovo p520.
:
: Tested-by: Kailang <kailang@realtek.com>
: Tested-by: Hui Wang <hui.wang@canonical.com>
: - conf/ucm: Rearrange Makefile.am
:
: Just rearrange and split to multiple lines to make future patches
: cleaner. No actual code change.
:
: - configure.ac: build extra mixer modules conditionally
:
: - Change snd_dlopen() function to return the error string
:
: The dlopen() function might fail also for another reason than
: a missing file, thus return the error string from dlerror().
:
===Documentation===
: - Change FSF address (Franklin Street)
:
===Dynamic Loader helpers===
: - core: Proper reference of internal snd_dlopen()
:
: snd_dlopen() was recently rewritten to be versioned symbols, and we
: have to call it with INTERNAL() wrapper from the library itself.
:
: Add the proper declaration in the local header and fix the callers
: appropriately.
:
===External PCM I/O Plugin SDK===
: - pcm: ioplug: Provide hw_avail helper function for plugins
:
: This function can be called without calling snd_pcm_avail_update().
:
: The call to snd_pcm_avail_update() can take some time.
: Therefore some developers would not like to call it from a real-time
: context (e.g. from JACK client context).
:
: - pcm: ioplug: Use boundary for wrap around
:
: if requested by the IO plugin
:
: Without this changes an IO plugin is not able to report
: that buffer_size frames were read from the buffer.
: When the buffer was full this is a valid action and
: has not to be handled as an under run.
:
: For example when the hw_ptr will be updated with
: hw_ptr += buffer_size
: and it is using the buffer_size as wrap around
: hw_ptr %= buffer_size
: would result in the same value as before the add operation.
:
===Kernel Headers===
: - asound.h: add SNDRV_PCM_FORMAT_{S, U}20
:
: This patch adds definitions of 20-bit, 4-byte PCM formats
: SNDRV_PCM_FORMAT_{S,U}20, that were recently added to the kernel, to
: alsa-lib's own copy of asound.h.
:
: Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
===Simple Abstraction Mixer Modules===
: - modules: smixer_python - add support for python3
:
: - configure.ac: build extra mixer modules conditionally
:
: - Change snd_dlopen() function to return the error string
:
: The dlopen() function might fail also for another reason than
: a missing file, thus return the error string from dlerror().
:
===Test/Example code===
: - test: correct emulation for channel-map TLV
:
: Current implementation of channel-map TLV on test program is not valid.
: Furthermore, it brings buffer-over-run due to byte counting.
:
: This commit fixes it.
:
==alsa-utils==
===Core===
: - Release v1.1.6
:
===ALSA Control (alsactl)===
: - alsactl: Only start restore service when asoundrc file exists
:
: This solves the chicken and egg problem on fresh installations whereby
: the alsa state file does not yet exist, and alsa-restore unit attempted
: to launch without first having a state file.
:
: - Change FSF address (Franklin Street)
:
===ALSA RawMidi Utility (amidi)===
: - Change FSF address (Franklin Street)
:
===Speaker Test===
: - speaker-test: Refactor the tone-generator codes
:
: There are many redundant open codes in speaker-test for performing the
: similar things, and especially the tone generator codes are ugly.
: Let's clean up a bit. This patch combines all open-codes into a
: single common helper with the callback for generating the tone.
:
===alsaloop===
: - alsaloop: fix a typo in the comparison
:
: Hello maintainers,
:
: I would like to suggest you a patch which fixes a typo in the alsa-utils
: alsaloop.
:
: Best Regards,
: Kirill Marinushkin
:
: - Change FSF address (Franklin Street)
:
===amixer===
: - Change FSF address (Franklin Street)
:
===aplay/arecord===
: - aplay: Fix wav file not being split on 32 bit platforms
:
: On my 32 bit armhf board arecord exits because of write() returning EFBIG
: when the output file size reaches 2147483647 bytes.
:
: To fix this, include generated header file before system header files
: so that _FILE_OFFSET_BITS=64 is used properly, as required in documentation
: "man feature_test_macros".
:
: - aplay: Adjust sample rate limits to support newer hardware
:
: There are number of devices that support up to 384 kHz sampling rate and
: some devices up to 768 kHz sampling rate. This patch increases sanity
: check limit to 768k in order to support testing of such hardware.
:
===bat (basic audio tester)===
: - bat: alsa.c - move the thread cleanup pop before goto exit3
:
==alsa-tools==
===Core===
: - Release v1.1.6
:
: - Makefile: fix version file creation
:
===Digigram Echo Mixer===
: - echomixer: add .desktop and icon files
:
: - Change FSF address (Franklin Street)
:
===Envy24 Control===
: - envy24control: add .desktop and icon files
:
: - Change FSF address (Franklin Street)
:
===HDA Jack Retask===
: - hdajackretask: add .desktop and icon files
:
===HDA Jack Sense Test===
: - hdajacksensetest: add gitcompile to EXTRA_DIST
:
: The gitcompile script is used from the top-level Makefile, so it needs
: to be included in the tarball.
:
===as10k1 (EMU10K1+ DSP Assembler)===
: - Change FSF address (Franklin Street)
:
===hdspconf===
: - hdspconf: move .desktop and icon files to desktop, use modern destination for icons
:
: - hdspconf, hdspmixer: validate the .desktop files
:
===hdspmixer===
: - hdspmixer: move .desktop and icon files to desktop, use modern destination for icons
:
: - hdspconf, hdspmixer: validate the .desktop files
:
===hwmixvolume===
: - hwmixvolume: add .desktop and icon files
:
: Resized version of public domain clipart found here:
: http://www.openclipart.org/detail/17428
:
: - hwmixvolume: use python2 directly
:
==alsa-plugins==
===Core===
: - Release v1.1.6
:
: - Change FSF address (Franklin Street)
:
===A52 Output plugin===
: - Change FSF address (Franklin Street)
:
===Arcam AV control plugin===
: - Change FSF address (Franklin Street)
:
===Jack PCM plugin===
: - jack: Use correct parameter order for snd_pcm_areas_silence()
:
: Without this fix it will sometimes fail with a segmentation fault.
:
: - jack: Report Xruns to user application
:
: Only increasing the hw_ptr is not sufficient
: because it will not be evaluated by the ALSA library
: to detect an Xrun.
:
: In addition there is a raise where an Xrun detected by the JACK thread
: could not be detected in the ALSA thread.
: - In playback use case
: - The hw_ptr will be increased by the JACK thread
: (hw_ptr > appl_ptr => Xrun)
: - But the ALSA thread increases the appl_ptr before evaluating the
: hw_ptr
: - Therefore the hw_ptr < appl_ptr again
: - ALSA will not detect the Xrun which was already detected by the
: JACK thread
:
: Therefore an additional variable is required to report an Xrun from the
: JACK thread to ALSA.
:
: - jack: Do not Xrun the ALSA buffer
:
: when the JACK thread is requesting too many audio frames
:
: Playback:
: Without this commit the ALSA audio buffer will be played with endless
: repeats as long as the user application has not provided new audio data.
: Therefore this garbage will be played as long as the user application has
: not called snd_pcm_stop() after an Xrun. With this fix the rest of the
: JACK buffer will be filled with silence.
:
: Capture:
: Without this commit the audio data in the ALSA buffer would be
: overwritten. With this commit the new data from the JACK buffer will not
: be copied. Therefore the existing data in the ALSA buffer will not be
: overwritten.
:
: - jack: Use boundary as hw_ptr wrap around
:
: instead of using buffer_size as wrap around.
:
: This is required to detect Xruns.
:
: It is also required to allow the JACK thread
: to processes the whole ALSA audio buffer at once
: without calling snd_pcm_avail_update() in between.
:
: For example when the hw_ptr will be updated with
: hw_ptr += buffer_size
: and it is using the buffer_size as wrap around
: hw_ptr %= buffer_size
: would result in the same value as before the add operation.
:
: Due to that the user application would not recognize
: that the complete audio buffer was copied.
:
: - jack: Write only valid values to the internal hw_ptr
:
: Otherwise the ALSA thread could for example read a hw_ptr >
: buffer_size
:
: - Change FSF address (Franklin Street)
:
==tinycompress==
===Core===
: - Release v1.1.6
:
===Utilities===
: - cplay: make get_codec_iec return void
:
: get_codec_iec() doesn't return anything, so make return type as void
:
: - cplay: support IEC61937 format
:
: This patch adds very simple supports for IEC61937 S/PDIF format.
:
: The cplay just specifies the format ID and sends audio data to ALSA.
: There is no check the audio data is valid or invalid as IEC61937
: specifications.
:
: - cplay: move around MP3 header parsing method
:
: This patch moves MP3 parsing method for -I option, this is for
: adding other codec ID in the future.
:
: - cplay: add option to specify codec ID
:
: This patch adds a -I command line option to set the codec ID,
: same as crecord.
:
: - cplay: fix size and nmemb of mp3 header of fread
:
: This patch swaps 'size' and 'nmemb' arguments of fread.
: The fread returns always 1 (because 'nmemb' is 1) so checking
: the return value is always failure.
:
=Detailed changelog between 1.0.29 and 1.1.6 releases=
=Changelog between 1.0.29 and 1.1.6 releases=
==alsa-python==
===Core===
: - Release v1.1.6
:
: - setup: add some missing fields, add header files from pyalsa to MANIFEST
:
: - setup.py: check for the existence of the build directory
:
: - common.h: fix InternFromString()
:
===Documentation===
: - Change FSF address (Franklin Street)
:
===Test python scripts===
: - hctltest2: use 'hw:0' device instead 'default'
:
===pyalsa.alsacard module===
: - initial python3 support
:
: - Change FSF address (Franklin Street)
:
===pyalsa.alsacontrol module===
: - fix tp_free() calls, remove wrong PyObject_GC_Del use
:
: - initial python3 support
:
: - Change FSF address (Franklin Street)
:
===pyalsa.alsahcontrol module===
: - fix tp_free() calls, remove wrong PyObject_GC_Del use
:
===pyalsa.alsaseq module===
: - alsaseq: simplify TCONSTADD
:

Detailed changes v1.1.5 v1.1.6 - Revision history

Perex: 1 revision

Perex at 06:54, 4 April 2018

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