Merge changes Iaf4cced5,I3f44eee1 into msm-3.0
* changes:
compress: add compress parameter definations
compress API documentation
diff --git a/Documentation/sound/alsa/compress/snd_compress_data.txt b/Documentation/sound/alsa/compress/snd_compress_data.txt
new file mode 100644
index 0000000..a8e6762
--- /dev/null
+++ b/Documentation/sound/alsa/compress/snd_compress_data.txt
@@ -0,0 +1,186 @@
+ snd_compress_data.txt
+ =====================
+ 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 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.
+
+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 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_codecs
+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.
+
+- 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 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.
+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)
+
+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
+
+Instead,
+ offloaded processing 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
+ mechanisms.
+
+- 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
+
+Credits:
+- Mark Brown and Liam Girdwood for discussions on the need for this 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/include/sound/snd_compress_params.h b/include/sound/snd_compress_params.h
new file mode 100644
index 0000000..7203e5f
--- /dev/null
+++ b/include/sound/snd_compress_params.h
@@ -0,0 +1,396 @@
+/*
+ * snd_compress_params.h - codec types and parameters for compressed data
+ * streaming interface
+ *
+ * Copyright (C) 2011 Intel Corporation
+ * Authors: Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
+ * Vinod Koul <vinod.koul@linux.intel.com>
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * The definitions in this file are derived from the OpenMAX AL version 1.1
+ * and OpenMAX IL v 1.1.2 header files which contain the copyright notice below.
+ *
+ * Copyright (c) 2007-2010 The Khronos Group Inc.
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and/or associated documentation files (the
+ * "Materials "), to deal in the Materials without restriction, including
+ * without limitation the rights to use, copy, modify, merge, publish,
+ * distribute, sublicense, and/or sell copies of the Materials, and to
+ * permit persons to whom the Materials are furnished to do so, subject to
+ * the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included
+ * in all copies or substantial portions of the Materials.
+ *
+ * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
+ * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
+ * MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS.
+ *
+ */
+
+
+/* AUDIO CODECS SUPPORTED */
+#define MAX_NUM_CODECS 32
+#define MAX_NUM_CODEC_DESCRIPTORS 32
+#define MAX_NUM_RATES 32
+#define MAX_NUM_BITRATES 32
+
+/* Codecs are listed linearly to allow for extensibility */
+#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001)
+#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002)
+#define SND_AUDIOCODEC_AMR ((__u32) 0x00000003)
+#define SND_AUDIOCODEC_AMRWB ((__u32) 0x00000004)
+#define SND_AUDIOCODEC_AMRWBPLUS ((__u32) 0x00000005)
+#define SND_AUDIOCODEC_AAC ((__u32) 0x00000006)
+#define SND_AUDIOCODEC_WMA ((__u32) 0x00000007)
+#define SND_AUDIOCODEC_REAL ((__u32) 0x00000008)
+#define SND_AUDIOCODEC_VORBIS ((__u32) 0x00000009)
+#define SND_AUDIOCODEC_FLAC ((__u32) 0x0000000A)
+#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B)
+
+/*
+ * Profile and modes are listed with bit masks. This allows for a
+ * more compact representation of fields that will not evolve
+ * (in contrast to the list of codecs)
+ */
+
+#define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001)
+
+/* MP3 modes are only useful for encoders */
+#define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001)
+#define SND_AUDIOCHANMODE_MP3_STEREO ((__u32) 0x00000002)
+#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO ((__u32) 0x00000004)
+#define SND_AUDIOCHANMODE_MP3_DUAL ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_AMR ((__u32) 0x00000001)
+
+/* AMR modes are only useful for encoders */
+#define SND_AUDIOMODE_AMR_DTX_OFF ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMR_VAD1 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMR_VAD2 ((__u32) 0x00000004)
+
+#define SND_AUDIOSTREAMFORMAT_UNDEFINED ((__u32) 0x00000000)
+#define SND_AUDIOSTREAMFORMAT_CONFORMANCE ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_IF1 ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_IF2 ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_FSF ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_ITU ((__u32) 0x00000020)
+
+#define SND_AUDIOPROFILE_AMRWB ((__u32) 0x00000001)
+
+/* AMRWB modes are only useful for encoders */
+#define SND_AUDIOMODE_AMRWB_DTX_OFF ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMRWB_VAD1 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMRWB_VAD2 ((__u32) 0x00000004)
+
+#define SND_AUDIOPROFILE_AMRWBPLUS ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_AAC ((__u32) 0x00000001)
+
+/* AAC modes are required for encoders and decoders */
+#define SND_AUDIOMODE_AAC_MAIN ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AAC_LC ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AAC_SSR ((__u32) 0x00000004)
+#define SND_AUDIOMODE_AAC_LTP ((__u32) 0x00000008)
+#define SND_AUDIOMODE_AAC_HE ((__u32) 0x00000010)
+#define SND_AUDIOMODE_AAC_SCALABLE ((__u32) 0x00000020)
+#define SND_AUDIOMODE_AAC_ERLC ((__u32) 0x00000040)
+#define SND_AUDIOMODE_AAC_LD ((__u32) 0x00000080)
+#define SND_AUDIOMODE_AAC_HE_PS ((__u32) 0x00000100)
+#define SND_AUDIOMODE_AAC_HE_MPS ((__u32) 0x00000200)
+
+/* AAC formats are required for encoders and decoders */
+#define SND_AUDIOSTREAMFORMAT_MP2ADTS ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_MP4ADTS ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_MP4LOAS ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_MP4LATM ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_ADIF ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_MP4FF ((__u32) 0x00000020)
+#define SND_AUDIOSTREAMFORMAT_RAW ((__u32) 0x00000040)
+
+#define SND_AUDIOPROFILE_WMA7 ((__u32) 0x00000001)
+#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002)
+#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004)
+#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008)
+
+#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001)
+#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_WMA_LEVEL3 ((__u32) 0x00000004)
+#define SND_AUDIOMODE_WMA_LEVEL4 ((__u32) 0x00000008)
+#define SND_AUDIOMODE_WMAPRO_LEVELM0 ((__u32) 0x00000010)
+#define SND_AUDIOMODE_WMAPRO_LEVELM1 ((__u32) 0x00000020)
+#define SND_AUDIOMODE_WMAPRO_LEVELM2 ((__u32) 0x00000040)
+#define SND_AUDIOMODE_WMAPRO_LEVELM3 ((__u32) 0x00000080)
+
+#define SND_AUDIOSTREAMFORMAT_WMA_ASF ((__u32) 0x00000001)
+/*
+ * Some implementations strip the ASF header and only send ASF packets
+ * to the DSP
+ */
+#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR ((__u32) 0x00000002)
+
+#define SND_AUDIOPROFILE_REALAUDIO ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_REALAUDIO_G2 ((__u32) 0x00000001)
+#define SND_AUDIOMODE_REALAUDIO_8 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_REALAUDIO_10 ((__u32) 0x00000004)
+#define SND_AUDIOMODE_REALAUDIO_SURROUND ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_VORBIS ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_VORBIS ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_FLAC ((__u32) 0x00000001)
+
+/*
+ * Define quality levels for FLAC encoders, from LEVEL0 (fast)
+ * to LEVEL8 (best)
+ */
+#define SND_AUDIOMODE_FLAC_LEVEL0 ((__u32) 0x00000001)
+#define SND_AUDIOMODE_FLAC_LEVEL1 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_FLAC_LEVEL2 ((__u32) 0x00000004)
+#define SND_AUDIOMODE_FLAC_LEVEL3 ((__u32) 0x00000008)
+#define SND_AUDIOMODE_FLAC_LEVEL4 ((__u32) 0x00000010)
+#define SND_AUDIOMODE_FLAC_LEVEL5 ((__u32) 0x00000020)
+#define SND_AUDIOMODE_FLAC_LEVEL6 ((__u32) 0x00000040)
+#define SND_AUDIOMODE_FLAC_LEVEL7 ((__u32) 0x00000080)
+#define SND_AUDIOMODE_FLAC_LEVEL8 ((__u32) 0x00000100)
+
+#define SND_AUDIOSTREAMFORMAT_FLAC ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_FLAC_OGG ((__u32) 0x00000002)
+
+/* IEC61937 payloads without CUVP and preambles */
+#define SND_AUDIOPROFILE_IEC61937 ((__u32) 0x00000001)
+/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */
+#define SND_AUDIOPROFILE_IEC61937_SPDIF ((__u32) 0x00000002)
+
+/*
+ * IEC modes are mandatory for decoders. Format autodetection
+ * will only happen on the DSP side with mode 0. The PCM mode should
+ * not be used, the PCM codec should be used instead
+ */
+#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER ((__u32) 0x00000000)
+#define SND_AUDIOMODE_IEC_LPCM ((__u32) 0x00000001)
+#define SND_AUDIOMODE_IEC_AC3 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_IEC_MPEG1 ((__u32) 0x00000004)
+#define SND_AUDIOMODE_IEC_MP3 ((__u32) 0x00000008)
+#define SND_AUDIOMODE_IEC_MPEG2 ((__u32) 0x00000010)
+#define SND_AUDIOMODE_IEC_AACLC ((__u32) 0x00000020)
+#define SND_AUDIOMODE_IEC_DTS ((__u32) 0x00000040)
+#define SND_AUDIOMODE_IEC_ATRAC ((__u32) 0x00000080)
+#define SND_AUDIOMODE_IEC_SACD ((__u32) 0x00000100)
+#define SND_AUDIOMODE_IEC_EAC3 ((__u32) 0x00000200)
+#define SND_AUDIOMODE_IEC_DTS_HD ((__u32) 0x00000400)
+#define SND_AUDIOMODE_IEC_MLP ((__u32) 0x00000800)
+#define SND_AUDIOMODE_IEC_DST ((__u32) 0x00001000)
+#define SND_AUDIOMODE_IEC_WMAPRO ((__u32) 0x00002000)
+#define SND_AUDIOMODE_IEC_REF_CXT ((__u32) 0x00004000)
+#define SND_AUDIOMODE_IEC_HE_AAC ((__u32) 0x00008000)
+#define SND_AUDIOMODE_IEC_HE_AAC2 ((__u32) 0x00010000)
+#define SND_AUDIOMODE_IEC_MPEG_SURROUND ((__u32) 0x00020000)
+
+/* <FIXME: multichannel encoders aren't supported for now. Would need
+ an additional definition of channel arrangement> */
+
+/* VBR/CBR definitions */
+#define SND_RATECONTROLMODE_CONSTANTBITRATE ((__u32) 0x00000001)
+#define SND_RATECONTROLMODE_VARIABLEBITRATE ((__u32) 0x00000002)
+
+/* Encoder options */
+
+struct wmaEncoderOptions {
+ __u32 super_block_align; /* WMA Type-specific data */
+};
+
+
+/**
+ * struct vorbisEncoderOptions
+ * @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
+ * In the default mode of operation, the quality level is 3.
+ * Normal quality range is 0 - 10.
+ * @managed: Boolean. Set bitrate management mode. This turns off the
+ * normal VBR encoding, but allows hard or soft bitrate constraints to be
+ * enforced by the encoder. This mode can be slower, and may also be
+ * lower quality. It is primarily useful for streaming.
+ * @maxBitrate: enabled only is managed is TRUE
+ * @minBitrate: enabled only is managed is TRUE
+ * @downmix: Boolean. Downmix input from stereo to mono (has no effect on
+ * non-stereo streams). Useful for lower-bitrate encoding.
+ *
+ * These options were extracted from the OpenMAX IL spec and gstreamer vorbisenc
+ * properties
+ *
+ * For best quality users should specify VBR mode and set quality levels.
+ */
+
+struct vorbisEncoderOptions {
+ int quality;
+ __u32 managed;
+ __u32 maxBitrate;
+ __u32 minBirate;
+ __u32 downmix;
+};
+
+
+/**
+ * struct realEncoderOptions
+ * @coupling_quant_bits: is the number of coupling quantization bits in the stream
+ * @coupling_start_region: is the coupling start region in the stream
+ * @num_regions: is the number of regions value
+ *
+ * These options were extracted from the OpenMAX IL spec
+ */
+
+struct realEncoderOptions {
+ __u32 coupling_quant_bits;
+ __u32 coupling_start_region;
+ __u32 num_regions;
+};
+
+/**
+ * struct flacEncoderOptions
+ * @serialNumber: valid only for OGG formats, needs to be set by application
+ * @replayGain: Add ReplayGain tags
+ *
+ * These options were extracted from the FLAC online documentation
+ * at http://flac.sourceforge.net/documentation_tools_flac.html
+ *
+ * To make the API simpler, it is assumed that the user will select quality
+ * profiles. Additional options that affect encoding quality and speed can
+ * be added at a later stage if need be.
+ *
+ * By default the Subset format is used by encoders.
+ *
+ * TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are
+ * not supported in this API.
+ */
+
+struct flacEncoderOptions {
+ __u32 serialNumber;
+ __u32 replayGain;
+};
+
+struct genericEncoderOptions {
+ __u32 encoderBandwidth;
+ int reserved[15];
+};
+
+union AudioCodecOptions {
+ struct wmaEncoderOptions wmaSpecificOptions;
+ struct vorbisEncoderOptions vorbisSpecificOptions;
+ struct realEncoderOptions realSpecificOptions;
+ struct flacEncoderOptions flacEncoderOptions;
+ struct genericEncoderOptions genericOptions;
+};
+
+/** struct SndAudioCodecDescriptor - description of codec capabilities
+ * @maxChannels: maximum number of audio channels
+ * @minBitsPerSample: Minimum bits per sample of PCM data <FIXME: needed?>
+ * @maxBitsPerSample: Maximum bits per sample of PCM data <FIXME: needed?>
+ * @minSampleRate: Minimum sampling rate supported, unit is Hz
+ * @maxSampleRate: Minimum sampling rate supported, unit is Hz
+ * @isFreqRangeContinuous: TRUE if the device supports a continuous range of
+ * sampling rates between minSampleRate and maxSampleRate;
+ * otherwise FALSE <FIXME: needed?>
+ * @SampleRatesSupported: Indexed array containing supported sampling rates in Hz
+ * @numSampleRatesSupported: Size of the pSamplesRatesSupported array
+ * @minBitRate: Minimum bitrate in bits per second
+ * @maxBitRate: Max bitrate in bits per second
+ * @isBitrateRangeContinuous: TRUE if the device supports a continuous range of
+ * bitrates between minBitRate and maxBitRate; otherwise FALSE
+ * @BitratesSupported: Indexed array containing supported bit rates
+ * @numBitratesSupported: Size of the pBiratesSupported array
+ * @rateControlSupported: value is specified by SND_RATECONTROLMODE defines.
+ * @profileSetting: Profile supported. See SND_AUDIOPROFILE defines.
+ * @modeSetting: Mode supported. See SND_AUDIOMODE defines
+ * @streamFormat: Format supported. See SND_AUDIOSTREAMFORMAT defines
+ * @reserved: reserved for future use
+ *
+ * This structure provides a scalar value for profile, mode and stream format fields.
+ * If an implementation supports multiple combinations, they will be listed as codecs
+ * with different IDs, for example there would be 2 decoders for AAC-RAW and AAC-ADTS.
+ * This entails some redundancy but makes it easier to avoid invalid configurations.
+ *
+ */
+
+struct SndAudioCodecDescriptor {
+ __u32 maxChannels;
+ __u32 minBitsPerSample;
+ __u32 maxBitsPerSample;
+ __u32 minSampleRate;
+ __u32 maxSampleRate;
+ __u32 isFreqRangeContinuous;
+ __u32 sampleRatesSupported[MAX_NUM_RATES];
+ __u32 numSampleRatesSupported;
+ __u32 minBitRate;
+ __u32 maxBitRate;
+ __u32 isBitrateRangeContinuous;
+ __u32 bitratesSupported[MAX_NUM_BITRATES];
+ __u32 numBitratesSupported;
+ __u32 rateControlSupported;
+ __u32 profileSetting;
+ __u32 modeSetting;
+ __u32 streamFormat;
+ __u32 reserved[16];
+};
+
+/** struct SndAudioCodecSettings -
+ * @codecId: Identifies the supported audio encoder/decoder. See SND_AUDIOCODEC macros.
+ * @channelsIn: Number of input audio channels
+ * @channelsOut: Number of output channels. In case of contradiction between this field and the
+ * channelMode field, the channelMode field overrides
+ * @sampleRate: Audio sample rate of input data
+ * @bitRate: Bitrate of encoded data. May be ignored by decoders
+ * @bitsPerSample: <FIXME: Needed? DSP implementations can handle their own format>
+ * @rateControl: Encoding rate control. See SND_RATECONTROLMODE defines.
+ * Encoders may rely on profiles for quality levels.
+ * May be ignored by decoders.
+ * @profileSetting: Mandatory for encoders, can be mandatory for specific decoders as well.
+ * See SND_AUDIOPROFILE defines
+ * @levelSetting: Supported level (Only used by WMA at the moment)
+ * @channelMode: Channel mode for encoder. See SND_AUDIOCHANMODE defines
+ * @streamFormat: Format of encoded bistream. Mandatory when defined. See SND_AUDIOSTREAMFORMAT
+ * defines
+ * @blockAlignment: Block alignment in bytes of an audio sample. Only required for PCM or IEC formats
+ * @options: encoder-specific settings
+ * @reserved: reserved for future use
+ */
+
+struct SndAudioCodecSettings {
+ __u32 codecId;
+ __u32 channelsIn;
+ __u32 channelsOut;
+ __u32 sampleRate;
+ __u32 bitRate;
+ __u32 bitsPerSample;
+ __u32 rateControl;
+ __u32 profileSetting;
+ __u32 levelSetting;
+ __u32 channelMode;
+ __u32 streamFormat;
+ __u32 blockAlignment;
+ union AudioCodecOptions options;
+ __u32 reserved[3];
+};