Diffstat (limited to 'Documentation/sound/alsa')
4 files changed, 196 insertions, 22 deletions
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 4f3443230d8..c8c54544abc 100644
@@ -42,19 +42,7 @@ ALC260
- fujitsu Fujitsu Laptop
- benq Benq ED8
- benq-t31 Benq T31
- hippo Hippo (ATI) with jack detection, Sony UX-90s
- hippo_1 Hippo (Benq) with jack detection
- toshiba-s06 Toshiba S06
- toshiba-rx1 Toshiba RX1
- tyan Tyan Thunder n6650W (S2915-E)
- ultra Samsung Q1 Ultra Vista model
- lenovo-3000 Lenovo 3000 y410
- nec NEC Versa S9100
- basic fixed pin assignment w/o SPDIF
- auto auto-config reading BIOS (default)
@@ -349,7 +337,7 @@ STAC92HD83*
ref Reference board
mic-ref Reference board with power management for ports
dell-s14 Dell laptop
- hp HP laptops with (inverted) mute-LED
+ dell-vostro-3500 Dell Vostro 3500 laptop
hp-dv7-4000 HP dv-7 4000
auto BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 03e2771ddee..91fee3b45fb 100644
@@ -579,7 +579,7 @@ Development Tree
The latest development codes for HD-audio are found on sound git tree:
The master branch or for-next branches can be used as the main
development branches in general while the HD-audio specific patches
@@ -594,7 +594,7 @@ is, installed via the usual spells: configure, make and make
install(-modules). See INSTALL in the package. The snapshot tarballs
are found at:
Sending a Bug Report
@@ -696,7 +696,7 @@ via hda-verb won't change the mixer value.
The hda-verb program is found in the ftp directory:
Also a git repository is available:
@@ -764,7 +764,7 @@ operation, the jack plugging simulation, etc.
The package is found in:
A git repository is available:
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
new file mode 100644
@@ -0,0 +1,188 @@
+ Pierre-Louis.Bossart <email@example.com>
+ Vinod Koul <firstname.lastname@example.org>
+Since its early days, the ALSA API was defined with PCM support or
+constant bitrates payloads such as IEC61937 in mind. Arguments and
+returned values in frames are the norm, making it a challenge to
+extend the existing API to compressed data streams.
+In recent years, audio digital signal processors (DSP) were integrated
+in system-on-chip designs, and DSPs are also integrated in audio
+codecs. Processing compressed data on such DSPs results in a dramatic
+reduction of power consumption compared to host-based
+processing. Support for such hardware has not been very good in Linux,
+mostly because of a lack of a generic API available in the mainline
+Rather than requiring a compability break with an API change of the
+ALSA PCM interface, a new 'Compressed Data' API is introduced to
+provide a control and data-streaming interface for audio DSPs.
+The design of this API was inspired by the 2-year experience with the
+Intel Moorestown SOC, with many corrections required to upstream the
+API in the mainline kernel instead of the staging tree and make it
+usable by others.
+The main requirements are:
+- separation between byte counts and time. Compressed formats may have
+ a header per file, per frame, or no header at all. The payload size
+ may vary from frame-to-frame. As a result, it is not possible to
+ estimate reliably the duration of audio buffers when handling
+ compressed data. Dedicated mechanisms are required to allow for
+ reliable audio-video synchronization, which requires precise
+ reporting of the number of samples rendered at any given time.
+- Handling of multiple formats. PCM data only requires a specification
+ of the sampling rate, number of channels and bits per sample. In
+ contrast, compressed data comes in a variety of formats. Audio DSPs
+ may also provide support for a limited number of audio encoders and
+ decoders embedded in firmware, or may support more choices through
+ dynamic download of libraries.
+- Focus on main formats. This API provides support for the most
+ popular formats used for audio and video capture and playback. It is
+ likely that as audio compression technology advances, new formats
+ will be added.
+- Handling of multiple configurations. Even for a given format like
+ AAC, some implementations may support AAC multichannel but HE-AAC
+ stereo. Likewise WMA10 level M3 may require too much memory and cpu
+ cycles. The new API needs to provide a generic way of listing these
+- Rendering/Grabbing only. This API does not provide any means of
+ hardware acceleration, where PCM samples are provided back to
+ user-space for additional processing. This API focuses instead on
+ streaming compressed data to a DSP, with the assumption that the
+ decoded samples are routed to a physical output or logical back-end.
+ - Complexity hiding. Existing user-space multimedia frameworks all
+ have existing enums/structures for each compressed format. This new
+ API assumes the existence of a platform-specific compatibility layer
+ to expose, translate and make use of the capabilities of the audio
+ DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
+ applications are not supposed to make use of this API.
+The new API shares a number of concepts with with the PCM API for flow
+control. Start, pause, resume, drain and stop commands have the same
+semantics no matter what the content is.
+The concept of memory ring buffer divided in a set of fragments is
+borrowed from the ALSA PCM API. However, only sizes in bytes can be
+Seeks/trick modes are assumed to be handled by the host.
+The notion of rewinds/forwards is not supported. Data committed to the
+ring buffer cannot be invalidated, except when dropping all buffers.
+The Compressed Data API does not make any assumptions on how the data
+is transmitted to the audio DSP. DMA transfers from main memory to an
+embedded audio cluster or to a SPI interface for external DSPs are
+possible. As in the ALSA PCM case, a core set of routines is exposed;
+each driver implementer will have to write support for a set of
+mandatory routines and possibly make use of optional ones.
+The main additions are
+This routine returns the list of audio formats supported. Querying the
+codecs on a capture stream will return encoders, decoders will be
+listed for playback streams.
+- get_codec_caps For each codec, this routine returns a list of
+capabilities. The intent is to make sure all the capabilities
+correspond to valid settings, and to minimize the risks of
+configuration failures. For example, for a complex codec such as AAC,
+the number of channels supported may depend on a specific profile. If
+the capabilities were exposed with a single descriptor, it may happen
+that a specific combination of profiles/channels/formats may not be
+supported. Likewise, embedded DSPs have limited memory and cpu cycles,
+it is likely that some implementations make the list of capabilities
+dynamic and dependent on existing workloads. In addition to codec
+settings, this routine returns the minimum buffer size handled by the
+implementation. This information can be a function of the DMA buffer
+sizes, the number of bytes required to synchronize, etc, and can be
+used by userspace to define how much needs to be written in the ring
+buffer before playback can start.
+This routine sets the configuration chosen for a specific codec. The
+most important field in the parameters is the codec type; in most
+cases decoders will ignore other fields, while encoders will strictly
+comply to the settings
+This routines returns the actual settings used by the DSP. Changes to
+the settings should remain the exception.
+The timestamp becomes a multiple field structure. It lists the number
+of bytes transferred, the number of samples processed and the number
+of samples rendered/grabbed. All these values can be used to determine
+the avarage bitrate, figure out if the ring buffer needs to be
+refilled or the delay due to decoding/encoding/io on the DSP.
+Note that the list of codecs/profiles/modes was derived from the
+OpenMAX AL specification instead of reinventing the wheel.
+- Addition of FLAC and IEC formats
+- Merge of encoder/decoder capabilities
+- Profiles/modes listed as bitmasks to make descriptors more compact
+- Addition of set_params for decoders (missing in OpenMAX AL)
+- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
+- Addition of format information for WMA
+- Addition of encoding options when required (derived from OpenMAX IL)
+- Addition of rateControlSupported (missing in OpenMAX AL)
+- Support for VoIP/circuit-switched calls is not the target of this
+ API. Support for dynamic bit-rate changes would require a tight
+ coupling between the DSP and the host stack, limiting power savings.
+- Packet-loss concealment is not supported. This would require an
+ additional interface to let the decoder synthesize data when frames
+ are lost during transmission. This may be added in the future.
+- Volume control/routing is not handled by this API. Devices exposing a
+ compressed data interface will be considered as regular ALSA devices;
+ volume changes and routing information will be provided with regular
+ ALSA kcontrols.
+- Embedded audio effects. Such effects should be enabled in the same
+ manner, no matter if the input was PCM or compressed.
+- multichannel IEC encoding. Unclear if this is required.
+- Encoding/decoding acceleration is not supported as mentioned
+ above. It is possible to route the output of a decoder to a capture
+ stream, or even implement transcoding capabilities. This routing
+ would be enabled with ALSA kcontrols.
+- Audio policy/resource management. This API does not provide any
+ hooks to query the utilization of the audio DSP, nor any premption
+- No notion of underun/overrun. Since the bytes written are compressed
+ in nature and data written/read doesn't translate directly to
+ rendered output in time, this does not deal with underrun/overun and
+ maybe dealt in user-library
+- Mark Brown and Liam Girdwood for discussions on the need for this API
+- Harsha Priya for her work on intel_sst compressed API
+- Rakesh Ughreja for valuable feedback
+- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
+ demonstrating and quantifying the benefits of audio offload on a
+ real platform.
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 3e2ec9cbf39..d50c14df341 100644
@@ -50,8 +50,7 @@ Machine DAI Configuration
The machine DAI configuration glues all the codec and CPU DAIs together. It can
also be used to set up the DAI system clock and for any machine related DAI
initialisation e.g. the machine audio map can be connected to the codec audio
-map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
+map, unconnected codec pins can be set as such.
struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
@@ -83,8 +82,7 @@ Machine Power Map
The machine driver can optionally extend the codec power map and to become an
audio power map of the audio subsystem. This allows for automatic power up/down
of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
-sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
+sockets in the machine init function.