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Based on kernel version 3.13. Page generated on 2014-01-20 22:04 EST.

1			compress_offload.txt
2			=====================
3		Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
4			Vinod Koul <vinod.koul@linux.intel.com>
6	Overview
8	Since its early days, the ALSA API was defined with PCM support or
9	constant bitrates payloads such as IEC61937 in mind. Arguments and
10	returned values in frames are the norm, making it a challenge to
11	extend the existing API to compressed data streams.
13	In recent years, audio digital signal processors (DSP) were integrated
14	in system-on-chip designs, and DSPs are also integrated in audio
15	codecs. Processing compressed data on such DSPs results in a dramatic
16	reduction of power consumption compared to host-based
17	processing. Support for such hardware has not been very good in Linux,
18	mostly because of a lack of a generic API available in the mainline
19	kernel.
21	Rather than requiring a compatibility break with an API change of the
22	ALSA PCM interface, a new 'Compressed Data' API is introduced to
23	provide a control and data-streaming interface for audio DSPs.
25	The design of this API was inspired by the 2-year experience with the
26	Intel Moorestown SOC, with many corrections required to upstream the
27	API in the mainline kernel instead of the staging tree and make it
28	usable by others.
30	Requirements
32	The main requirements are:
34	- separation between byte counts and time. Compressed formats may have
35	  a header per file, per frame, or no header at all. The payload size
36	  may vary from frame-to-frame. As a result, it is not possible to
37	  estimate reliably the duration of audio buffers when handling
38	  compressed data. Dedicated mechanisms are required to allow for
39	  reliable audio-video synchronization, which requires precise
40	  reporting of the number of samples rendered at any given time.
42	- Handling of multiple formats. PCM data only requires a specification
43	  of the sampling rate, number of channels and bits per sample. In
44	  contrast, compressed data comes in a variety of formats. Audio DSPs
45	  may also provide support for a limited number of audio encoders and
46	  decoders embedded in firmware, or may support more choices through
47	  dynamic download of libraries.
49	- Focus on main formats. This API provides support for the most
50	  popular formats used for audio and video capture and playback. It is
51	  likely that as audio compression technology advances, new formats
52	  will be added.
54	- Handling of multiple configurations. Even for a given format like
55	  AAC, some implementations may support AAC multichannel but HE-AAC
56	  stereo. Likewise WMA10 level M3 may require too much memory and cpu
57	  cycles. The new API needs to provide a generic way of listing these
58	  formats.
60	- Rendering/Grabbing only. This API does not provide any means of
61	  hardware acceleration, where PCM samples are provided back to
62	  user-space for additional processing. This API focuses instead on
63	  streaming compressed data to a DSP, with the assumption that the
64	  decoded samples are routed to a physical output or logical back-end.
66	 - Complexity hiding. Existing user-space multimedia frameworks all
67	  have existing enums/structures for each compressed format. This new
68	  API assumes the existence of a platform-specific compatibility layer
69	  to expose, translate and make use of the capabilities of the audio
70	  DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
71	  applications are not supposed to make use of this API.
74	Design
76	The new API shares a number of concepts with the PCM API for flow
77	control. Start, pause, resume, drain and stop commands have the same
78	semantics no matter what the content is.
80	The concept of memory ring buffer divided in a set of fragments is
81	borrowed from the ALSA PCM API. However, only sizes in bytes can be
82	specified.
84	Seeks/trick modes are assumed to be handled by the host.
86	The notion of rewinds/forwards is not supported. Data committed to the
87	ring buffer cannot be invalidated, except when dropping all buffers.
89	The Compressed Data API does not make any assumptions on how the data
90	is transmitted to the audio DSP. DMA transfers from main memory to an
91	embedded audio cluster or to a SPI interface for external DSPs are
92	possible. As in the ALSA PCM case, a core set of routines is exposed;
93	each driver implementer will have to write support for a set of
94	mandatory routines and possibly make use of optional ones.
96	The main additions are
98	- get_caps
99	This routine returns the list of audio formats supported. Querying the
100	codecs on a capture stream will return encoders, decoders will be
101	listed for playback streams.
103	- get_codec_caps For each codec, this routine returns a list of
104	capabilities. The intent is to make sure all the capabilities
105	correspond to valid settings, and to minimize the risks of
106	configuration failures. For example, for a complex codec such as AAC,
107	the number of channels supported may depend on a specific profile. If
108	the capabilities were exposed with a single descriptor, it may happen
109	that a specific combination of profiles/channels/formats may not be
110	supported. Likewise, embedded DSPs have limited memory and cpu cycles,
111	it is likely that some implementations make the list of capabilities
112	dynamic and dependent on existing workloads. In addition to codec
113	settings, this routine returns the minimum buffer size handled by the
114	implementation. This information can be a function of the DMA buffer
115	sizes, the number of bytes required to synchronize, etc, and can be
116	used by userspace to define how much needs to be written in the ring
117	buffer before playback can start.
119	- set_params
120	This routine sets the configuration chosen for a specific codec. The
121	most important field in the parameters is the codec type; in most
122	cases decoders will ignore other fields, while encoders will strictly
123	comply to the settings
125	- get_params
126	This routines returns the actual settings used by the DSP. Changes to
127	the settings should remain the exception.
129	- get_timestamp
130	The timestamp becomes a multiple field structure. It lists the number
131	of bytes transferred, the number of samples processed and the number
132	of samples rendered/grabbed. All these values can be used to determine
133	the average bitrate, figure out if the ring buffer needs to be
134	refilled or the delay due to decoding/encoding/io on the DSP.
136	Note that the list of codecs/profiles/modes was derived from the
137	OpenMAX AL specification instead of reinventing the wheel.
138	Modifications include:
139	- Addition of FLAC and IEC formats
140	- Merge of encoder/decoder capabilities
141	- Profiles/modes listed as bitmasks to make descriptors more compact
142	- Addition of set_params for decoders (missing in OpenMAX AL)
143	- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
144	- Addition of format information for WMA
145	- Addition of encoding options when required (derived from OpenMAX IL)
146	- Addition of rateControlSupported (missing in OpenMAX AL)
148	Gapless Playback
149	================
150	When playing thru an album, the decoders have the ability to skip the encoder
151	delay and padding and directly move from one track content to another. The end
152	user can perceive this as gapless playback as we dont have silence while
153	switching from one track to another
155	Also, there might be low-intensity noises due to encoding. Perfect gapless is
156	difficult to reach with all types of compressed data, but works fine with most
157	music content. The decoder needs to know the encoder delay and encoder padding.
158	So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
159	and are not present by default in the bitstream, hence the need for a new
160	interface to pass this information to the DSP. Also DSP and userspace needs to
161	switch from one track to another and start using data for second track.
163	The main additions are:
165	- set_metadata
166	This routine sets the encoder delay and encoder padding. This can be used by
167	decoder to strip the silence. This needs to be set before the data in the track
168	is written.
170	- set_next_track
171	This routine tells DSP that metadata and write operation sent after this would
172	correspond to subsequent track
174	- partial drain
175	This is called when end of file is reached. The userspace can inform DSP that
176	EOF is reached and now DSP can start skipping padding delay. Also next write
177	data would belong to next track
179	Sequence flow for gapless would be:
180	- Open
181	- Get caps / codec caps
182	- Set params
183	- Set metadata of the first track
184	- Fill data of the first track
185	- Trigger start
186	- User-space finished sending all,
187	- Indicaite next track data by sending set_next_track
188	- Set metadata of the next track
189	- then call partial_drain to flush most of buffer in DSP
190	- Fill data of the next track
191	- DSP switches to second track
192	(note: order for partial_drain and write for next track can be reversed as well)
194	Not supported:
196	- Support for VoIP/circuit-switched calls is not the target of this
197	  API. Support for dynamic bit-rate changes would require a tight
198	  coupling between the DSP and the host stack, limiting power savings.
200	- Packet-loss concealment is not supported. This would require an
201	  additional interface to let the decoder synthesize data when frames
202	  are lost during transmission. This may be added in the future.
204	- Volume control/routing is not handled by this API. Devices exposing a
205	  compressed data interface will be considered as regular ALSA devices;
206	  volume changes and routing information will be provided with regular
207	  ALSA kcontrols.
209	- Embedded audio effects. Such effects should be enabled in the same
210	  manner, no matter if the input was PCM or compressed.
212	- multichannel IEC encoding. Unclear if this is required.
214	- Encoding/decoding acceleration is not supported as mentioned
215	  above. It is possible to route the output of a decoder to a capture
216	  stream, or even implement transcoding capabilities. This routing
217	  would be enabled with ALSA kcontrols.
219	- Audio policy/resource management. This API does not provide any
220	  hooks to query the utilization of the audio DSP, nor any preemption
221	  mechanisms.
223	- No notion of underrun/overrun. Since the bytes written are compressed
224	  in nature and data written/read doesn't translate directly to
225	  rendered output in time, this does not deal with underrun/overrun and
226	  maybe dealt in user-library
228	Credits:
229	- Mark Brown and Liam Girdwood for discussions on the need for this API
230	- Harsha Priya for her work on intel_sst compressed API
231	- Rakesh Ughreja for valuable feedback
232	- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
233	  demonstrating and quantifying the benefits of audio offload on a
234	  real platform.
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