329 lines
15 KiB
ReStructuredText
329 lines
15 KiB
ReStructuredText
=========================
|
|
ALSA Compress-Offload API
|
|
=========================
|
|
|
|
Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
|
|
|
|
Vinod Koul <vinod.koul@linux.intel.com>
|
|
|
|
|
|
Overview
|
|
========
|
|
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
|
|
kernel.
|
|
|
|
Rather than requiring a compatibility 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.
|
|
|
|
|
|
Requirements
|
|
============
|
|
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
|
|
formats.
|
|
|
|
- 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.
|
|
|
|
|
|
Design
|
|
======
|
|
The new API shares a number of concepts 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
|
|
specified.
|
|
|
|
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
|
|
|
|
get_caps
|
|
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.
|
|
|
|
set_params
|
|
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
|
|
|
|
get_params
|
|
This routines returns the actual settings used by the DSP. Changes to
|
|
the settings should remain the exception.
|
|
|
|
get_timestamp
|
|
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 average 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.
|
|
Modifications include:
|
|
- 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)
|
|
|
|
State Machine
|
|
=============
|
|
|
|
The compressed audio stream state machine is described below ::
|
|
|
|
+----------+
|
|
| |
|
|
| OPEN |
|
|
| |
|
|
+----------+
|
|
|
|
|
|
|
|
| compr_set_params()
|
|
|
|
|
v
|
|
compr_free() +----------+
|
|
+------------------------------------| |
|
|
| | SETUP |
|
|
| +-------------------------| |<-------------------------+
|
|
| | compr_write() +----------+ |
|
|
| | ^ |
|
|
| | | compr_drain_notify() |
|
|
| | | or |
|
|
| | | compr_stop() |
|
|
| | | |
|
|
| | +----------+ |
|
|
| | | | |
|
|
| | | DRAIN | |
|
|
| | | | |
|
|
| | +----------+ |
|
|
| | ^ |
|
|
| | | |
|
|
| | | compr_drain() |
|
|
| | | |
|
|
| v | |
|
|
| +----------+ +----------+ |
|
|
| | | compr_start() | | compr_stop() |
|
|
| | PREPARE |------------------->| RUNNING |--------------------------+
|
|
| | | | | |
|
|
| +----------+ +----------+ |
|
|
| | | ^ |
|
|
| |compr_free() | | |
|
|
| | compr_pause() | | compr_resume() |
|
|
| | | | |
|
|
| v v | |
|
|
| +----------+ +----------+ |
|
|
| | | | | compr_stop() |
|
|
+--->| FREE | | PAUSE |---------------------------+
|
|
| | | |
|
|
+----------+ +----------+
|
|
|
|
|
|
Gapless Playback
|
|
================
|
|
When playing thru an album, the decoders have the ability to skip the encoder
|
|
delay and padding and directly move from one track content to another. The end
|
|
user can perceive this as gapless playback as we don't have silence while
|
|
switching from one track to another
|
|
|
|
Also, there might be low-intensity noises due to encoding. Perfect gapless is
|
|
difficult to reach with all types of compressed data, but works fine with most
|
|
music content. The decoder needs to know the encoder delay and encoder padding.
|
|
So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
|
|
and are not present by default in the bitstream, hence the need for a new
|
|
interface to pass this information to the DSP. Also DSP and userspace needs to
|
|
switch from one track to another and start using data for second track.
|
|
|
|
The main additions are:
|
|
|
|
set_metadata
|
|
This routine sets the encoder delay and encoder padding. This can be used by
|
|
decoder to strip the silence. This needs to be set before the data in the track
|
|
is written.
|
|
|
|
set_next_track
|
|
This routine tells DSP that metadata and write operation sent after this would
|
|
correspond to subsequent track
|
|
|
|
partial drain
|
|
This is called when end of file is reached. The userspace can inform DSP that
|
|
EOF is reached and now DSP can start skipping padding delay. Also next write
|
|
data would belong to next track
|
|
|
|
Sequence flow for gapless would be:
|
|
- Open
|
|
- Get caps / codec caps
|
|
- Set params
|
|
- Set metadata of the first track
|
|
- Fill data of the first track
|
|
- Trigger start
|
|
- User-space finished sending all,
|
|
- Indicate next track data by sending set_next_track
|
|
- Set metadata of the next track
|
|
- then call partial_drain to flush most of buffer in DSP
|
|
- Fill data of the next track
|
|
- DSP switches to second track
|
|
|
|
(note: order for partial_drain and write for next track can be reversed as well)
|
|
|
|
Gapless Playback SM
|
|
===================
|
|
|
|
For Gapless, we move from running state to partial drain and back, along
|
|
with setting of meta_data and signalling for next track ::
|
|
|
|
|
|
+----------+
|
|
compr_drain_notify() | |
|
|
+------------------------>| RUNNING |
|
|
| | |
|
|
| +----------+
|
|
| |
|
|
| |
|
|
| | compr_next_track()
|
|
| |
|
|
| V
|
|
| +----------+
|
|
| | |
|
|
| |NEXT_TRACK|
|
|
| | |
|
|
| +----------+
|
|
| |
|
|
| |
|
|
| | compr_partial_drain()
|
|
| |
|
|
| V
|
|
| +----------+
|
|
| | |
|
|
+------------------------ | PARTIAL_ |
|
|
| DRAIN |
|
|
+----------+
|
|
|
|
Not supported
|
|
=============
|
|
- 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 preemption
|
|
mechanisms.
|
|
|
|
- No notion of underrun/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/overrun and
|
|
maybe dealt in user-library
|
|
|
|
|
|
Credits
|
|
=======
|
|
- 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.
|