252 lines
10 KiB
ReStructuredText
252 lines
10 KiB
ReStructuredText
=========================
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ALSA Compress-Offload API
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=========================
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Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
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Vinod Koul <vinod.koul@linux.intel.com>
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Overview
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========
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Since its early days, the ALSA API was defined with PCM support or
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constant bitrates payloads such as IEC61937 in mind. Arguments and
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returned values in frames are the norm, making it a challenge to
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extend the existing API to compressed data streams.
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In recent years, audio digital signal processors (DSP) were integrated
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in system-on-chip designs, and DSPs are also integrated in audio
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codecs. Processing compressed data on such DSPs results in a dramatic
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reduction of power consumption compared to host-based
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processing. Support for such hardware has not been very good in Linux,
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mostly because of a lack of a generic API available in the mainline
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kernel.
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Rather than requiring a compatibility break with an API change of the
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ALSA PCM interface, a new 'Compressed Data' API is introduced to
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provide a control and data-streaming interface for audio DSPs.
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The design of this API was inspired by the 2-year experience with the
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Intel Moorestown SOC, with many corrections required to upstream the
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API in the mainline kernel instead of the staging tree and make it
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usable by others.
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Requirements
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============
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The main requirements are:
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- separation between byte counts and time. Compressed formats may have
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a header per file, per frame, or no header at all. The payload size
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may vary from frame-to-frame. As a result, it is not possible to
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estimate reliably the duration of audio buffers when handling
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compressed data. Dedicated mechanisms are required to allow for
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reliable audio-video synchronization, which requires precise
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reporting of the number of samples rendered at any given time.
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- Handling of multiple formats. PCM data only requires a specification
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of the sampling rate, number of channels and bits per sample. In
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contrast, compressed data comes in a variety of formats. Audio DSPs
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may also provide support for a limited number of audio encoders and
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decoders embedded in firmware, or may support more choices through
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dynamic download of libraries.
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- Focus on main formats. This API provides support for the most
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popular formats used for audio and video capture and playback. It is
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likely that as audio compression technology advances, new formats
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will be added.
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- Handling of multiple configurations. Even for a given format like
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AAC, some implementations may support AAC multichannel but HE-AAC
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stereo. Likewise WMA10 level M3 may require too much memory and cpu
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cycles. The new API needs to provide a generic way of listing these
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formats.
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- Rendering/Grabbing only. This API does not provide any means of
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hardware acceleration, where PCM samples are provided back to
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user-space for additional processing. This API focuses instead on
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streaming compressed data to a DSP, with the assumption that the
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decoded samples are routed to a physical output or logical back-end.
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- Complexity hiding. Existing user-space multimedia frameworks all
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have existing enums/structures for each compressed format. This new
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API assumes the existence of a platform-specific compatibility layer
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to expose, translate and make use of the capabilities of the audio
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DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
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applications are not supposed to make use of this API.
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Design
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======
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The new API shares a number of concepts with the PCM API for flow
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control. Start, pause, resume, drain and stop commands have the same
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semantics no matter what the content is.
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The concept of memory ring buffer divided in a set of fragments is
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borrowed from the ALSA PCM API. However, only sizes in bytes can be
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specified.
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Seeks/trick modes are assumed to be handled by the host.
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The notion of rewinds/forwards is not supported. Data committed to the
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ring buffer cannot be invalidated, except when dropping all buffers.
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The Compressed Data API does not make any assumptions on how the data
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is transmitted to the audio DSP. DMA transfers from main memory to an
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embedded audio cluster or to a SPI interface for external DSPs are
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possible. As in the ALSA PCM case, a core set of routines is exposed;
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each driver implementer will have to write support for a set of
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mandatory routines and possibly make use of optional ones.
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The main additions are
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get_caps
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This routine returns the list of audio formats supported. Querying the
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codecs on a capture stream will return encoders, decoders will be
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listed for playback streams.
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get_codec_caps
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For each codec, this routine returns a list of
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capabilities. The intent is to make sure all the capabilities
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correspond to valid settings, and to minimize the risks of
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configuration failures. For example, for a complex codec such as AAC,
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the number of channels supported may depend on a specific profile. If
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the capabilities were exposed with a single descriptor, it may happen
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that a specific combination of profiles/channels/formats may not be
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supported. Likewise, embedded DSPs have limited memory and cpu cycles,
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it is likely that some implementations make the list of capabilities
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dynamic and dependent on existing workloads. In addition to codec
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settings, this routine returns the minimum buffer size handled by the
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implementation. This information can be a function of the DMA buffer
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sizes, the number of bytes required to synchronize, etc, and can be
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used by userspace to define how much needs to be written in the ring
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buffer before playback can start.
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set_params
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This routine sets the configuration chosen for a specific codec. The
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most important field in the parameters is the codec type; in most
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cases decoders will ignore other fields, while encoders will strictly
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comply to the settings
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get_params
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This routines returns the actual settings used by the DSP. Changes to
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the settings should remain the exception.
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get_timestamp
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The timestamp becomes a multiple field structure. It lists the number
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of bytes transferred, the number of samples processed and the number
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of samples rendered/grabbed. All these values can be used to determine
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the average bitrate, figure out if the ring buffer needs to be
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refilled or the delay due to decoding/encoding/io on the DSP.
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Note that the list of codecs/profiles/modes was derived from the
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OpenMAX AL specification instead of reinventing the wheel.
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Modifications include:
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- Addition of FLAC and IEC formats
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- Merge of encoder/decoder capabilities
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- Profiles/modes listed as bitmasks to make descriptors more compact
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- Addition of set_params for decoders (missing in OpenMAX AL)
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- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
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- Addition of format information for WMA
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- Addition of encoding options when required (derived from OpenMAX IL)
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- Addition of rateControlSupported (missing in OpenMAX AL)
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Gapless Playback
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================
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When playing thru an album, the decoders have the ability to skip the encoder
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delay and padding and directly move from one track content to another. The end
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user can perceive this as gapless playback as we don't have silence while
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switching from one track to another
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Also, there might be low-intensity noises due to encoding. Perfect gapless is
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difficult to reach with all types of compressed data, but works fine with most
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music content. The decoder needs to know the encoder delay and encoder padding.
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So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
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and are not present by default in the bitstream, hence the need for a new
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interface to pass this information to the DSP. Also DSP and userspace needs to
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switch from one track to another and start using data for second track.
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The main additions are:
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set_metadata
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This routine sets the encoder delay and encoder padding. This can be used by
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decoder to strip the silence. This needs to be set before the data in the track
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is written.
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set_next_track
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This routine tells DSP that metadata and write operation sent after this would
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correspond to subsequent track
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partial drain
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This is called when end of file is reached. The userspace can inform DSP that
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EOF is reached and now DSP can start skipping padding delay. Also next write
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data would belong to next track
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- set_next_track_param
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This routine is called to send to DSP codec specific data of subsequent track
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in gapless before first write.
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Sequence flow for gapless would be:
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- Open
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- Get caps / codec caps
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- Set params
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- Set metadata of the first track
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- Fill data of the first track
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- Trigger start
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- User-space finished sending all,
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- Indicate next track data by sending set_next_track
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- Set metadata of the next track
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- then call partial_drain to flush most of buffer in DSP
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- set codec specific data of subsequent track
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- Fill data of the next track
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- DSP switches to second track
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(note: order for partial_drain and write for next track can be reversed as well)
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Not supported
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=============
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- Support for VoIP/circuit-switched calls is not the target of this
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API. Support for dynamic bit-rate changes would require a tight
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coupling between the DSP and the host stack, limiting power savings.
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- Packet-loss concealment is not supported. This would require an
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additional interface to let the decoder synthesize data when frames
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are lost during transmission. This may be added in the future.
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- Volume control/routing is not handled by this API. Devices exposing a
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compressed data interface will be considered as regular ALSA devices;
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volume changes and routing information will be provided with regular
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ALSA kcontrols.
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- Embedded audio effects. Such effects should be enabled in the same
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manner, no matter if the input was PCM or compressed.
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- multichannel IEC encoding. Unclear if this is required.
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- Encoding/decoding acceleration is not supported as mentioned
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above. It is possible to route the output of a decoder to a capture
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stream, or even implement transcoding capabilities. This routing
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would be enabled with ALSA kcontrols.
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- Audio policy/resource management. This API does not provide any
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hooks to query the utilization of the audio DSP, nor any preemption
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mechanisms.
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- No notion of underrun/overrun. Since the bytes written are compressed
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in nature and data written/read doesn't translate directly to
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rendered output in time, this does not deal with underrun/overrun and
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maybe dealt in user-library
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Credits
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=======
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- Mark Brown and Liam Girdwood for discussions on the need for this API
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- Harsha Priya for her work on intel_sst compressed API
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- Rakesh Ughreja for valuable feedback
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- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
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demonstrating and quantifying the benefits of audio offload on a
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real platform.
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