Am Freitag, den 02.09.2011, 11:36 +0530 schrieb Vinod Koul:

The patch adds the various definations used to define the encoder and decoder parameters

There is a typo in the summary and message body: defin*i*tions.

I found some more things in the comments below.

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

---

include/sound/snd_compress_params.h | 396 +++++++++++++++++++++++++++++++++++ 1 files changed, 396 insertions(+), 0 deletions(-) create mode 100644 include/sound/snd_compress_params.h

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

No ».« between the names.

• •          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

The alignment should be fixed. Should a full stop ».« be added at the end in all the comments?

• */

+#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

s/is/if/

• • @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

GStreamer, since the other words are also capitalized?

• • 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

Remove »is the«.

• • @coupling_start_region: is the coupling start region in the stream

Dito.

• • @num_regions: is the number of regions value

Dito.

• • 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.

s/if need be/if needed/

• • 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?>

More elaborate explanation for the FIXME (in the commit message)?

• • @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?>
    

Dito.

• • @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

Remove `p` in front of pBiratesSupported?

• • @rateControlSupported: value is specified by SND_RATECONTROLMODE defines.

*V*alue.

• • @profileSetting: Profile supported. See SND_AUDIOPROFILE defines.

Supported profile.

• • @modeSetting: Mode supported. See SND_AUDIOMODE defines

Supported mod.

• • @streamFormat: Format supported. See SND_AUDIOSTREAMFORMAT defines

Supported format.

• • @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.

White space before 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
    

Overrides what?

• • @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.
    

Alignment.

• • @profileSetting: Mandatory for encoders, can be mandatory for specific decoders as well.

• • See SND_AUDIOPROFILE defines
    

Alignement?

• • @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
    

Alignment?

• • @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];

+};

On Wed, 2011-09-07 at 16:20 +0100, Dimitris Papastamos wrote:

On Fri, Sep 02, 2011 at 11:36:24AM +0530, Vinod Koul wrote:

...

This patch ads core.c, the file which implements the ioctls and registers the devices

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

---

sound/compress_offload/core.c | 632 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 632 insertions(+), 0 deletions(-) create mode 100644 sound/compress_offload/core.c

diff --git a/sound/compress_offload/core.c b/sound/compress_offload/core.c new file mode 100644 index 0000000..1797dba --- /dev/null +++ b/sound/compress_offload/core.c @@ -0,0 +1,632 @@

...

+static int snd_compr_start(struct snd_compr_stream *stream) +{

• int retval;
• if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)

• return -EPERM;
  

• retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
• if (!retval)

• stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
  

• return retval;

+}

+static int snd_compr_stop(struct snd_compr_stream *stream) +{

• int retval;
• if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)

• return -EPERM;
  

• retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
• if (!retval) {

• stream->runtime->state = SNDRV_PCM_STATE_SETUP;
  

• wake_up(&stream->runtime->sleep);
  

• }
• return retval;

+}

+static int snd_compr_drain(struct snd_compr_stream *stream) +{

• int retval;
• if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED ||

• 	stream->runtime->state != SNDRV_PCM_STATE_PAUSED)
  

• return -EPERM;
  

• retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
• if (!retval) {

• stream->runtime->state = SNDRV_PCM_STATE_SETUP;
  

• wake_up(&stream->runtime->sleep);
  

• }
• return retval;

+}

How is it expected to have stream->runtime->state == SNDRV_PCM_STATE_PREPARED? This is not set anywhere in the core code. Do we need to just add another ioctl for _PREPARE?

Nope. My thoughts on this were that _TRIGGER_START should be called only when first data has been written, i.e when we are in _SETUP state. So write in _SETUP state should move this to _PREPARED, and obviously I have missed that part, Thanks will fix this

On Thu, 2011-09-08 at 11:38 +0100, Dimitris Papastamos wrote:

On Fri, Sep 02, 2011 at 11:36:24AM +0530, Vinod Koul wrote:

...

This patch ads core.c, the file which implements the ioctls and registers the devices

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

---

sound/compress_offload/core.c | 632 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 632 insertions(+), 0 deletions(-) create mode 100644 sound/compress_offload/core.c

diff --git a/sound/compress_offload/core.c b/sound/compress_offload/core.c new file mode 100644 index 0000000..1797dba --- /dev/null +++ b/sound/compress_offload/core.c @@ -0,0 +1,632 @@

...

+unsigned int snd_compr_poll(struct file *f, poll_table *wait) +{

• struct snd_ioctl_data *data = f->private_data;
• struct snd_compr_stream *stream;
• BUG_ON(!data);
• stream = &data->stream;
• if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)

• return -ENXIO;
  

• poll_wait(f, &stream->runtime->sleep, wait);
• /* this would change after read is implemented, we would need to

• * check for direction here */
  

• if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)

• return POLLOUT | POLLWRNORM;
  

• return 0;

+}

Do we need to grab any locks here? I'm also confused about checking for != SNDRV_PCM_STATE_RUNNING and only then returning POLLOUT | POLLWRNORM. Shouldn't the normal cycle be write->start->poll->write->poll->...? At the moment, the state won't change and for this cycle to be effective we have to call stop/drain in between. Am I missing something?

My original intention (and in staging sst driver), was state changes when poll is woken up, but I agree in above case it is not true. I will fix this

From: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

The patch adds the documentation file explaining the API

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- Documentation/sound/alsa/snd_compress_data.txt | 184 ++++++++++++++++++++++++ 1 files changed, 184 insertions(+), 0 deletions(-) create mode 100644 Documentation/sound/alsa/snd_compress_data.txt

diff --git a/Documentation/sound/alsa/snd_compress_data.txt b/Documentation/sound/alsa/snd_compress_data.txt new file mode 100644 index 0000000..4b0d0fc --- /dev/null +++ b/Documentation/sound/alsa/snd_compress_data.txt @@ -0,0 +1,184 @@ + 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_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. + +- 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; + 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 +- 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.

At Tue, 13 Dec 2011 14:32:56 +0530, Vinod Koul wrote:

From: Omair Mohammed Abdullah omair.m.abdullah@intel.com

Use the two minor numbers 2 and 3 for audio compressed offload devices. Also add support for these devices in core

Signed-off-by: Omair Mohammed Abdullah omair.m.abdullah@intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Reviewed-by: Mark Brown broonie@opensource.wolfsonmicro.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com

---

include/sound/core.h | 1 + include/sound/minors.h | 7 ++++++- sound/core/sound.c | 6 ++++++ 3 files changed, 13 insertions(+), 1 deletions(-)

diff --git a/include/sound/core.h b/include/sound/core.h index 3be5ab7..5ab255f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -62,6 +62,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) #define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) +#define SNDRV_DEV_COMPRESS ((__force snd_device_type_t) 0x100A) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000)

typedef int __bitwise snd_device_state_t; diff --git a/include/sound/minors.h b/include/sound/minors.h index 8f76420..faaeb64 100644 --- a/include/sound/minors.h +++ b/include/sound/minors.h @@ -35,7 +35,8 @@ #define SNDRV_MINOR_TIMER 33 /* SNDRV_MINOR_GLOBAL + 1 * 32 */

#ifndef CONFIG_SND_DYNAMIC_MINORS

• 				/* 2 - 3 (reserved) */
  

+#define SNDRV_MINOR_COMPR_PLAYBACK 2 +#define SNDRV_MINOR_COMPR_CAPTURE 3

If you'd share both playback and capture to a single device (as you already did in patch 5/6), no need to provide distinct numbers here. This would be simply

#define SNDRV_MINOR_COMPR 2

and define the single SNDRV_DEVICE_TYPE_COMPR. Then you'll have to up to two devices for the static minors and more for dynamic minors. So, the below would be:

diff --git a/sound/core/sound.c b/sound/core/sound.c index 828af35..2b680dd 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -233,6 +233,12 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev) return -EINVAL; minor = SNDRV_MINOR(card->number, type + dev); break;

• case SNDRV_DEVICE_TYPE_COMPR_PLAYBACK:
• case SNDRV_DEVICE_TYPE_COMPR_CAPTURE:

• if (snd_BUG_ON(!card))
  

• 	return -EINVAL;
  

• minor = SNDRV_MINOR(card->number, type);
  

• break;
  

Instead of this, just add case SNDRV_DEVICE_TYPE_COMPR in the cases of other types like PCM and RAWMIDI, i.e.

• minor = SNDRV_MINOR(card->number, type + dev);
  

Takashi

The patch adds the various definations used to define the encoder and decoder parameters

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Reviewed-by: Mark Brown broonie@opensource.wolfsonmicro.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- include/sound/snd_compress_params.h | 395 +++++++++++++++++++++++++++++++++++ 1 files changed, 395 insertions(+), 0 deletions(-) create mode 100644 include/sound/snd_compress_params.h

diff --git a/include/sound/snd_compress_params.h b/include/sound/snd_compress_params.h new file mode 100644 index 0000000..79b681d --- /dev/null +++ b/include/sound/snd_compress_params.h @@ -0,0 +1,395 @@ +/* + * 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. + * + */ +#ifndef __SND_COMPRESS_PARAMS_H +#define __SND_COMPRESS_PARAMS_H + +/* AUDIO CODECS SUPPORTED */ +#define MAX_NUM_CODECS 32 +#define MAX_NUM_CODEC_DESCRIPTORS 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) +#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) +#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) + +/* + * 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) + +#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002) +#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002) + +/* <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 snd_enc_wma { + __u32 super_block_align; /* WMA Type-specific data */ +}; + + +/** + * struct snd_enc_vorbis + * @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. + * @max_bit_rate: Enabled only if managed is TRUE + * @min_bit_rate: Enabled only if 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 snd_enc_vorbis { + int quality; + __u32 managed; + __u32 max_bit_rate; + __u32 min_bit_rate; + __u32 downmix; +}; + + +/** + * struct snd_enc_real + * @quant_bits: number of coupling quantization bits in the stream + * @start_region: coupling start region in the stream + * @num_regions: number of regions value + * + * These options were extracted from the OpenMAX IL spec + */ + +struct snd_enc_real { + __u32 quant_bits; + __u32 start_region; + __u32 num_regions; +}; + +/** + * struct snd_enc_flac + * @num: serial number, valid only for OGG formats + * needs to be set by application + * @gain: Add replay gain 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 needed. + * + * 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 snd_enc_flac { + __u32 num; + __u32 gain; +}; + +struct snd_enc_generic { + __u32 bw; /* encoder bandwidth */ + int reserved[15]; +}; + +union snd_codec_options { + struct snd_enc_wma wma; + struct snd_enc_vorbis vorbis; + struct snd_enc_real real; + struct snd_enc_flac flac; + struct snd_enc_generic generic; +}; + +/** struct snd_codec_desc - description of codec capabilities + * @max_ch: Maximum number of audio channels + * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this + * @bit_rate: Indexed array containing supported bit rates + * @num_bitrates: Number of valid values in bit_rate array + * @rate_control: value is specified by SND_RATECONTROLMODE defines. + * @profiles: Supported profiles. See SND_AUDIOPROFILE defines. + * @modes: Supported modes. See SND_AUDIOMODE defines + * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines + * @reserved: reserved for future use + * + * This structure provides a scalar value for profiles, modes and stream + * format fields. + * If an implementation supports multiple combinations, they will be listed as + * codecs with different descriptors, for example there would be 2 descriptors + * for AAC-RAW and AAC-ADTS. + * This entails some redundancy but makes it easier to avoid invalid + * configurations. + * + */ + +struct snd_codec_desc { + __u32 max_ch; + __u32 sample_rates; + __u32 bit_rate[MAX_NUM_BITRATES]; + __u32 num_bitrates; + __u32 rate_control; + __u32 profiles; + __u32 modes; + __u32 formats; + __u32 reserved[16]; +}; + +/** struct snd_codec + * @id: Identifies the supported audio encoder/decoder. + * See SND_AUDIOCODEC macros. + * @ch_in: Number of input audio channels + * @ch_out: Number of output channels. In case of contradiction between + * this field and the channelMode field, the channelMode field + * overrides. + * @sample_rate: Audio sample rate of input data + * @bit_rate: Bitrate of encoded data. May be ignored by decoders + * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines. + * Encoders may rely on profiles for quality levels. + * May be ignored by decoders. + * @profile: Mandatory for encoders, can be mandatory for specific + * decoders as well. See SND_AUDIOPROFILE defines. + * @level: Supported level (Only used by WMA at the moment) + * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines + * @format: Format of encoded bistream. Mandatory when defined. + * See SND_AUDIOSTREAMFORMAT defines. + * @align: 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 snd_codec { + __u32 id; + __u32 ch_in; + __u32 ch_out; + __u32 sample_rate; + __u32 bit_rate; + __u32 rate_control; + __u32 profile; + __u32 level; + __u32 ch_mode; + __u32 format; + __u32 align; + union snd_codec_options options; + __u32 reserved[3]; +}; + +#endif

This patch adds core.c, the file which implements the ioctls and registers the devices

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- sound/core/compress_offload.c | 705 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 705 insertions(+), 0 deletions(-) create mode 100644 sound/core/compress_offload.c

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c new file mode 100644 index 0000000..d2345c2 --- /dev/null +++ b/sound/core/compress_offload.c @@ -0,0 +1,705 @@ +/* + * core.c - compress offload core + * + * Copyright (C) 2011 Intel Corporation + * Authors: Vinod Koul vinod.koul@linux.intel.com + * Pierre-Louis Bossart pierre-louis.bossart@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#define FORMAT(fmt) "%s: %d: " fmt, __func__, __LINE__ +#define pr_fmt(fmt) KBUILD_MODNAME ": " FORMAT(fmt) + +#include <linux/file.h> +#include <linux/fs.h> +#include <linux/list.h> +#include <linux/mm.h> +#include <linux/mutex.h> +#include <linux/poll.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <linux/uio.h> +#include <linux/uaccess.h> +#include <linux/module.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/snd_compress_params.h> +#include <sound/compress_offload.h> +#include <sound/compress_driver.h> + +/* TODO: + * - add substream support for multiple devices in case of + * SND_DYNAMIC_MINORS is not used + * - Multiple node representation + * driver should be able to register multiple nodes + */ + +static DEFINE_MUTEX(device_mutex); + +struct snd_compr_file { + unsigned long caps; + struct snd_compr_stream stream; +}; + +static int snd_compr_open(struct inode *inode, struct file *f) +{ + struct snd_compr *compr; + struct snd_compr_file *data; + struct snd_compr_runtime *runtime; + int ret, type; + int maj = imajor(inode); + + if (f->f_flags & O_WRONLY) + type = SNDRV_DEVICE_TYPE_COMPR_PLAYBACK; + else if (f->f_flags & O_RDONLY) + type = SNDRV_DEVICE_TYPE_COMPR_CAPTURE; + else { + pr_err("invalid direction\n"); + return -EINVAL; + } + + if (maj == snd_major) + compr = snd_lookup_minor_data(iminor(inode), type); + else + return -ENXIO; + + if (compr == NULL) { + pr_err("no device data!!!\n"); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + data->stream.ops = compr->ops; + data->stream.direction = type; + data->stream.private_data = compr->private_data; + data->stream.device = compr; + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) { + kfree(data); + return -ENOMEM; + } + runtime->state = SNDRV_PCM_STATE_OPEN; + init_waitqueue_head(&runtime->sleep); + data->stream.runtime = runtime; + f->private_data = (void *)data; + mutex_lock(&compr->lock); + ret = compr->ops->open(&data->stream); + mutex_unlock(&compr->lock); + if (ret) { + kfree(runtime); + kfree(data); + } + return ret; +} + +static int snd_compr_free(struct inode *inode, struct file *f) +{ + struct snd_compr_file *data = f->private_data; + data->stream.ops->free(&data->stream); + kfree(data->stream.runtime->buffer); + kfree(data->stream.runtime); + kfree(data); + return 0; +} + +static void snd_compr_update_tstamp(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + if (!stream->ops->pointer) + return; + stream->ops->pointer(stream, tstamp); + pr_debug("dsp consumed till %ld total %ld bytes\n", + (long)tstamp->copied_bytes, (long)tstamp->copied_total); + stream->runtime->hw_pointer = tstamp->copied_bytes; + stream->runtime->bytes_copied = tstamp->copied_total; +} + +static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, + struct snd_compr_avail *avail) +{ + long avail_calc; /*this needs to be signed variable */ + + snd_compr_update_tstamp(stream, &avail->tstamp); + + if (stream->runtime->bytes_written == 0 && + stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + pr_debug("detected init and someone forgot to do a write\n"); + return stream->runtime->buffer_size; + } + pr_debug("app wrote %ld, DSP consumed %ld\n", + (long)stream->runtime->bytes_written, + (long)stream->runtime->bytes_copied); + if (stream->runtime->bytes_written == stream->runtime->bytes_copied) { + pr_debug("both pointers are same, returning full avail\n"); + return stream->runtime->buffer_size; + } + + avail_calc = stream->runtime->buffer_size - + (stream->runtime->app_pointer - stream->runtime->hw_pointer); + pr_debug("calc avail as %ld, app_ptr %ld, hw+ptr %ld\n", avail_calc, + (long)stream->runtime->app_pointer, + (long)stream->runtime->hw_pointer); + if (avail_calc >= stream->runtime->buffer_size) + avail_calc -= stream->runtime->buffer_size; + pr_debug("ret avail as %ld\n", avail_calc); + avail->avail = avail_calc; + return avail_calc; +} + +static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) +{ + struct snd_compr_avail avail; + + return snd_compr_calc_avail(stream, &avail); +} + +static int +snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_avail ioctl_avail; + size_t avail; + + avail = snd_compr_calc_avail(stream, &ioctl_avail); + ioctl_avail.avail = avail; + + if (copy_to_user((unsigned long __user *)arg, + &ioctl_avail, sizeof(ioctl_avail))) + return -EFAULT; + return 0; +} + +static int snd_compr_write_data(struct snd_compr_stream *stream, + const char __user *buf, size_t count) +{ + void *dstn; + size_t copy; + struct snd_compr_runtime *runtime = stream->runtime; + + dstn = runtime->buffer + runtime->app_pointer; + pr_debug("copying %ld at %ld\n", + (long)count, (long)runtime->app_pointer); + if (count < runtime->buffer_size - runtime->app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + runtime->app_pointer += count; + } else { + copy = runtime->buffer_size - runtime->app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(runtime->buffer, buf + copy, count - copy)) + return -EFAULT; + runtime->app_pointer = count - copy; + } + /* if DSP cares, let it know data has been written */ + if (stream->ops->ack) + stream->ops->ack(stream, count); + return count; +} + +static ssize_t snd_compr_write(struct file *f, const char __user *buf, + size_t count, loff_t *offset) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval; + + BUG_ON(!data); + stream = &data->stream; + mutex_lock(&stream->device->lock); + /* write is allowed when stream is running or has been steup */ + if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && + stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + mutex_unlock(&stream->device->lock); + return -EPERM; + } + + avail = snd_compr_get_avail(stream); + pr_debug("avail returned %ld\n", (long)avail); + /* calculate how much we can write to buffer */ + if (avail > count) + avail = count; + + if (stream->ops->copy) + retval = stream->ops->copy(stream, buf, avail); + else + retval = snd_compr_write_data(stream, buf, avail); + if (retval > 0) + stream->runtime->bytes_written += retval; + + /* while initiating the stream, write should be called before START + * call, so in setup move state */ + if (stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + pr_debug("stream prepared, Houston we are good to go\n"); + } + + mutex_unlock(&stream->device->lock); + return retval; +} + + +static ssize_t snd_compr_read(struct file *f, char __user *buf, + size_t count, loff_t *offset) +{ + return -ENXIO; +} + +static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) +{ + return -ENXIO; +} + +static unsigned int snd_compr_poll(struct file *f, poll_table *wait) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval = 0; + + BUG_ON(!data); + stream = &data->stream; + + mutex_lock(&stream->device->lock); + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + retval = -ENXIO; + goto out; + } + poll_wait(f, &stream->runtime->sleep, wait); + + avail = snd_compr_get_avail(stream); + pr_debug("avail is %ld\n", (long)avail); + /* check if we have at least one fragment to fill */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + if (avail >= stream->runtime->fragment_size) { + if (stream->direction == SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + retval = POLLOUT | POLLWRNORM; + else + retval = POLLIN | POLLRDNORM; + break; + } + /* Fall through */ + case SNDRV_PCM_STATE_DRAINING: + break; + default: + if (stream->direction == SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + retval = POLLOUT | POLLWRNORM | POLLERR; + else + retval = POLLIN | POLLRDNORM | POLLERR; + break; + } +out: + mutex_unlock(&stream->device->lock); + return retval; +} + +static int +snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_caps caps; + + if (!stream->ops->get_caps) + return -ENXIO; + + retval = stream->ops->get_caps(stream, &caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, &caps, sizeof(caps))) + retval = -EFAULT; +out: + return retval; +} + +static int +snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_codec_caps *caps; + + if (!stream->ops->get_codec_caps) + return -ENXIO; + + caps = kmalloc(sizeof(*caps), GFP_KERNEL); + if (!caps) + return -ENOMEM; + + retval = stream->ops->get_codec_caps(stream, caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, caps, sizeof(*caps))) + retval = -EFAULT; + +out: + kfree(caps); + return retval; +} + +/* revisit this with snd_pcm_preallocate_xxx */ +static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + unsigned int buffer_size; + void *buffer; + + buffer_size = params->buffer.fragment_size * params->buffer.fragments; + if (stream->ops->copy) { + buffer = NULL; + /* if copy is defined the driver will be required to copy + * the data from core + */ + } else { + buffer = kmalloc(buffer_size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + } + stream->runtime->fragment_size = params->buffer.fragment_size; + stream->runtime->fragments = params->buffer.fragments; + stream->runtime->buffer = buffer; + stream->runtime->buffer_size = buffer_size; + return 0; +} + +static int +snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_params *params; + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) + return -EFAULT; + retval = snd_compr_allocate_buffer(stream, params); + if (retval) { + kfree(params); + return -ENOMEM; + } + retval = stream->ops->set_params(stream, params); + if (retval) + goto out; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + } else + return -EPERM; +out: + kfree(params); + return retval; +} + +static int +snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->get_params) + return -ENXIO; + + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + retval = stream->ops->get_params(stream, params); + if (retval) + goto out; + if (copy_to_user((char __user *)arg, params, sizeof(*params))) + retval = -EFAULT; + +out: + kfree(params); + return retval; +} + +static inline int +snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_tstamp tstamp; + + snd_compr_update_tstamp(stream, &tstamp); + return copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; +} + +static int snd_compr_pause(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_resume(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_start(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_stop(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_drain(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + int retval = -ENOTTY; + + BUG_ON(!data); + stream = &data->stream; + BUG_ON(!stream); + mutex_lock(&stream->device->lock); + switch (_IOC_NR(cmd)) { + case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): + put_user(SNDRV_COMPRESS_VERSION, + (int __user *)arg) ? -EFAULT : 0; + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): + retval = snd_compr_get_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS): + retval = snd_compr_get_codec_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): + retval = snd_compr_set_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): + retval = snd_compr_get_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_TSTAMP): + retval = snd_compr_tstamp(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_AVAIL): + retval = snd_compr_ioctl_avail(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_PAUSE): + retval = snd_compr_pause(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_RESUME): + retval = snd_compr_resume(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_START): + retval = snd_compr_start(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_STOP): + retval = snd_compr_stop(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_DRAIN): + retval = snd_compr_drain(stream); + break; + } + mutex_unlock(&stream->device->lock); + return retval; +} + +static const struct file_operations snd_compr_file_ops = { + .owner = THIS_MODULE, + .open = snd_compr_open, + .release = snd_compr_free, + .write = snd_compr_write, + .read = snd_compr_read, + .unlocked_ioctl = snd_compr_ioctl, + .mmap = snd_compr_mmap, + .poll = snd_compr_poll, +}; + +static int snd_compress_dev_register(struct snd_device *device) +{ + int ret = -EINVAL; + char str[16]; + struct snd_compr *compr; + + if (snd_BUG_ON(!device || !device->device_data)) + return -ENXIO; + compr = device->device_data; + + sprintf(str, "comprC%iD%i", compr->card->number, compr->device); + pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + compr->direction); + /* register compressed device */ + ret = snd_register_device(compr->direction, compr->card, compr->device, + &snd_compr_file_ops, compr, str); + if (ret < 0) { + pr_err("snd_register_device failed\n %d", ret); + return ret; + } + return ret; + +} + +static int snd_compress_dev_disconnect(struct snd_device *device) +{ + struct snd_compr *compr; + + compr = device->device_data; + snd_unregister_device(compr->direction, compr->card, compr->device); + return 0; +} + +int snd_compress_new(struct snd_card *card, int device, + int type, struct snd_compr *compr) +{ + static struct snd_device_ops ops = { + .dev_free = NULL, + .dev_register = snd_compress_dev_register, + .dev_disconnect = snd_compress_dev_disconnect, + }; + + compr->card = card; + compr->device = device; + compr->direction = type; + return snd_device_new(card, SNDRV_DEV_COMPRESS, compr, &ops); +} +EXPORT_SYMBOL_GPL(snd_compress_new); + +static int snd_compress_add_device(struct snd_compr *device) +{ + int ret; + + if (!device->card) + return -EINVAL; + + /* register the card */ + ret = snd_card_register(device->card); + if (ret) + goto out; + return 0; + +out: + pr_err("failed with %d\n", ret); + return ret; + +} + +static int snd_compress_remove_device(struct snd_compr *device) +{ + return snd_card_free(device->card); +} + +/** + * snd_compress_register - register compressed device + * + * @device: compressed device to register + */ +int snd_compress_register(struct snd_compr *device) +{ + int retval; + + if (device->name == NULL || device->dev == NULL || device->ops == NULL) + return -EINVAL; + + pr_debug("Registering compressed device %s\n", device->name); + BUG_ON(!device->ops->open); + BUG_ON(!device->ops->free); + BUG_ON(!device->ops->set_params); + BUG_ON(!device->ops->trigger); + + mutex_init(&device->lock); + + /* register a compressed card */ + mutex_lock(&device_mutex); + retval = snd_compress_add_device(device); + mutex_unlock(&device_mutex); + return retval; +} +EXPORT_SYMBOL_GPL(snd_compress_register); + +int snd_compress_deregister(struct snd_compr *device) +{ + pr_debug("Removing compressed device %s\n", device->name); + mutex_lock(&device_mutex); + snd_compress_remove_device(device); + mutex_unlock(&device_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_compress_deregister); + +static int __init snd_compress_init(void) +{ + return 0; +} + +static void __exit snd_compress_exit(void) +{ +} + +module_init(snd_compress_init); +module_exit(snd_compress_exit); + +MODULE_DESCRIPTION("ALSA Compressed offload framework"); +MODULE_AUTHOR("Vinod Koul vinod.koul@linux.intel.com"); +MODULE_LICENSE("GPL v2");

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- sound/core/Kconfig | 8 ++++++++ sound/core/Makefile | 4 ++++ 2 files changed, 12 insertions(+), 0 deletions(-)

diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 475455c..5bfce50 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -155,6 +155,14 @@ config SND_DYNAMIC_MINORS

If you are unsure about this, say N here.

+config SND_COMPRESS_OFFLOAD + tristate "ALSA Compressed audio offload support" + default n + help + If you want support for offloading compressed audio and have such + a hardware, then you should say Y here and also to the DSP driver + of your platform. + config SND_SUPPORT_OLD_API bool "Support old ALSA API" default y diff --git a/sound/core/Makefile b/sound/core/Makefile index 350a08d..67c8e93 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -21,6 +21,8 @@ snd-hrtimer-objs := hrtimer.o snd-rtctimer-objs := rtctimer.o snd-hwdep-objs := hwdep.o

+snd-compress-objs := compress_offload.o + obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o @@ -31,3 +33,5 @@ obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o

obj-$(CONFIG_SND_OSSEMUL) += oss/ obj-$(CONFIG_SND_SEQUENCER) += seq/ + +obj-$(CONFIG_SND_COMPRESS_OFFLOAD) += snd-compress.o

From: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

The patch adds the documentation file explaining the API

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- Documentation/sound/alsa/compress_offload.txt | 188 +++++++++++++++++++++++++ 1 files changed, 188 insertions(+), 0 deletions(-) create mode 100644 Documentation/sound/alsa/compress_offload.txt

diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt new file mode 100644 index 0000000..c83a835 --- /dev/null +++ b/Documentation/sound/alsa/compress_offload.txt @@ -0,0 +1,188 @@ + compress_offload.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_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 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; + 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 +- 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.

The patch adds the various definations used to define the encoder and decoder parameters

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Reviewed-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/compress_params.h | 397 +++++++++++++++++++++++++++++++++++++++ 1 files changed, 397 insertions(+), 0 deletions(-) create mode 100644 include/sound/compress_params.h

diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h new file mode 100644 index 0000000..d97d69f --- /dev/null +++ b/include/sound/compress_params.h @@ -0,0 +1,397 @@ +/* + * 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. + * + */ +#ifndef __SND_COMPRESS_PARAMS_H +#define __SND_COMPRESS_PARAMS_H + +/* AUDIO CODECS SUPPORTED */ +#define MAX_NUM_CODECS 32 +#define MAX_NUM_CODEC_DESCRIPTORS 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) +#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) +#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) + +/* + * 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) + +#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002) +#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002) + +/* <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 snd_enc_wma { + __u32 super_block_align; /* WMA Type-specific data */ +}; + + +/** + * struct snd_enc_vorbis + * @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. + * @max_bit_rate: Enabled only if managed is TRUE + * @min_bit_rate: Enabled only if 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 snd_enc_vorbis { + __s32 quality; + __u32 managed; + __u32 max_bit_rate; + __u32 min_bit_rate; + __u32 downmix; +}; + + +/** + * struct snd_enc_real + * @quant_bits: number of coupling quantization bits in the stream + * @start_region: coupling start region in the stream + * @num_regions: number of regions value + * + * These options were extracted from the OpenMAX IL spec + */ + +struct snd_enc_real { + __u32 quant_bits; + __u32 start_region; + __u32 num_regions; +}; + +/** + * struct snd_enc_flac + * @num: serial number, valid only for OGG formats + * needs to be set by application + * @gain: Add replay gain 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 needed. + * + * 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 snd_enc_flac { + __u32 num; + __u32 gain; +}; + +struct snd_enc_generic { + __u32 bw; /* encoder bandwidth */ + __s32 reserved[15]; +}; + +union snd_codec_options { + struct snd_enc_wma wma; + struct snd_enc_vorbis vorbis; + struct snd_enc_real real; + struct snd_enc_flac flac; + struct snd_enc_generic generic; +}; + +/** struct snd_codec_desc - description of codec capabilities + * @max_ch: Maximum number of audio channels + * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this + * @bit_rate: Indexed array containing supported bit rates + * @num_bitrates: Number of valid values in bit_rate array + * @rate_control: value is specified by SND_RATECONTROLMODE defines. + * @profiles: Supported profiles. See SND_AUDIOPROFILE defines. + * @modes: Supported modes. See SND_AUDIOMODE defines + * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines + * @min_buffer: Minimum buffer size handled by codec implementation + * @reserved: reserved for future use + * + * This structure provides a scalar value for profiles, modes and stream + * format fields. + * If an implementation supports multiple combinations, they will be listed as + * codecs with different descriptors, for example there would be 2 descriptors + * for AAC-RAW and AAC-ADTS. + * This entails some redundancy but makes it easier to avoid invalid + * configurations. + * + */ + +struct snd_codec_desc { + __u32 max_ch; + __u32 sample_rates; + __u32 bit_rate[MAX_NUM_BITRATES]; + __u32 num_bitrates; + __u32 rate_control; + __u32 profiles; + __u32 modes; + __u32 formats; + __u32 min_buffer; + __u32 reserved[15]; +}; + +/** struct snd_codec + * @id: Identifies the supported audio encoder/decoder. + * See SND_AUDIOCODEC macros. + * @ch_in: Number of input audio channels + * @ch_out: Number of output channels. In case of contradiction between + * this field and the channelMode field, the channelMode field + * overrides. + * @sample_rate: Audio sample rate of input data + * @bit_rate: Bitrate of encoded data. May be ignored by decoders + * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines. + * Encoders may rely on profiles for quality levels. + * May be ignored by decoders. + * @profile: Mandatory for encoders, can be mandatory for specific + * decoders as well. See SND_AUDIOPROFILE defines. + * @level: Supported level (Only used by WMA at the moment) + * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines + * @format: Format of encoded bistream. Mandatory when defined. + * See SND_AUDIOSTREAMFORMAT defines. + * @align: 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 snd_codec { + __u32 id; + __u32 ch_in; + __u32 ch_out; + __u32 sample_rate; + __u32 bit_rate; + __u32 rate_control; + __u32 profile; + __u32 level; + __u32 ch_mode; + __u32 format; + __u32 align; + union snd_codec_options options; + __u32 reserved[3]; +}; + +#endif

This patch adds core.c, the file which implements the ioctls and registers the devices

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/core/compress_offload.c | 765 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 765 insertions(+), 0 deletions(-) create mode 100644 sound/core/compress_offload.c

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c new file mode 100644 index 0000000..dac3633 --- /dev/null +++ b/sound/core/compress_offload.c @@ -0,0 +1,765 @@ +/* + * compress_core.c - compress offload core + * + * Copyright (C) 2011 Intel Corporation + * Authors: Vinod Koul vinod.koul@linux.intel.com + * Pierre-Louis Bossart pierre-louis.bossart@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#define FORMAT(fmt) "%s: %d: " fmt, __func__, __LINE__ +#define pr_fmt(fmt) KBUILD_MODNAME ": " FORMAT(fmt) + +#include <linux/file.h> +#include <linux/fs.h> +#include <linux/list.h> +#include <linux/mm.h> +#include <linux/mutex.h> +#include <linux/poll.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <linux/uio.h> +#include <linux/uaccess.h> +#include <linux/module.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/compress_params.h> +#include <sound/compress_offload.h> +#include <sound/compress_driver.h> + +/* TODO: + * - add substream support for multiple devices in case of + * SND_DYNAMIC_MINORS is not used + * - Multiple node representation + * driver should be able to register multiple nodes + */ + +static DEFINE_MUTEX(device_mutex); + +struct snd_compr_file { + unsigned long caps; + struct snd_compr_stream stream; +}; + +/* + * a note on stream states used: + * we use follwing states in the compressed core + * SNDRV_PCM_STATE_OPEN: When stream has been opened. + * SNDRV_PCM_STATE_SETUP: When stream has been initialized. This is done by + * calling SNDRV_COMPRESS_SET_PARAMS. running streams will come to this + * state at stop by calling SNDRV_COMPRESS_STOP, or at end of drain. + * SNDRV_PCM_STATE_RUNNING: When stream has been started and is + * decoding/encoding and rendering/capturing data. + * SNDRV_PCM_STATE_DRAINING: When stream is draining current data. This is done + * by calling SNDRV_COMPRESS_DRAIN. + * SNDRV_PCM_STATE_PAUSED: When stream is paused. This is done by calling + * SNDRV_COMPRESS_PAUSE. It can be stopped or resumed by calling + * SNDRV_COMPRESS_STOP or SNDRV_COMPRESS_RESUME respectively. + */ +static int snd_compr_open(struct inode *inode, struct file *f) +{ + struct snd_compr *compr; + struct snd_compr_file *data; + struct snd_compr_runtime *runtime; + enum snd_compr_direction dirn; + int maj = imajor(inode); + int ret; + + if (f->f_flags & O_WRONLY) + dirn = SND_COMPRESS_PLAYBACK; + else if (f->f_flags & O_RDONLY) + dirn = SND_COMPRESS_CAPTURE; + else { + pr_err("invalid direction\n"); + return -EINVAL; + } + + if (maj == snd_major) + compr = snd_lookup_minor_data(iminor(inode), + SNDRV_DEVICE_TYPE_COMPRESS); + else + return -EBADFD; + + if (compr == NULL) { + pr_err("no device data!!!\n"); + return -ENODEV; + } + + if (dirn != compr->direction) { + pr_err("this device doesn't support this direction\n"); + return -EINVAL; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + data->stream.ops = compr->ops; + data->stream.direction = dirn; + data->stream.private_data = compr->private_data; + data->stream.device = compr; + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) { + kfree(data); + return -ENOMEM; + } + runtime->state = SNDRV_PCM_STATE_OPEN; + init_waitqueue_head(&runtime->sleep); + data->stream.runtime = runtime; + f->private_data = (void *)data; + mutex_lock(&compr->lock); + ret = compr->ops->open(&data->stream); + mutex_unlock(&compr->lock); + if (ret) { + kfree(runtime); + kfree(data); + } + return ret; +} + +static int snd_compr_free(struct inode *inode, struct file *f) +{ + struct snd_compr_file *data = f->private_data; + data->stream.ops->free(&data->stream); + kfree(data->stream.runtime->buffer); + kfree(data->stream.runtime); + kfree(data); + return 0; +} + +static void snd_compr_update_tstamp(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + if (!stream->ops->pointer) + return; + stream->ops->pointer(stream, tstamp); + pr_debug("dsp consumed till %d total %d bytes\n", + tstamp->byte_offset, tstamp->copied_total); + stream->runtime->hw_pointer = tstamp->byte_offset; + stream->runtime->total_bytes_transferred = tstamp->copied_total; +} + +static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, + struct snd_compr_avail *avail) +{ + long avail_calc; /*this needs to be signed variable */ + + snd_compr_update_tstamp(stream, &avail->tstamp); + + /* FIXME: This needs to be different for capture stream, + available is # of compressed data, for playback it's + remainder of buffer */ + + if (stream->runtime->total_bytes_available == 0 && + stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + pr_debug("detected init and someone forgot to do a write\n"); + return stream->runtime->buffer_size; + } + pr_debug("app wrote %lld, DSP consumed %lld\n", + stream->runtime->total_bytes_available, + stream->runtime->total_bytes_transferred); + if (stream->runtime->total_bytes_available == + stream->runtime->total_bytes_transferred) { + pr_debug("both pointers are same, returning full avail\n"); + return stream->runtime->buffer_size; + } + + /* FIXME: this routine isn't consistent, in one test we use + * cumulative values and in the other byte offsets. Do we + * really need the byte offsets if the cumulative values have + * been updated? In the PCM interface app_ptr and hw_ptr are + * already cumulative */ + + avail_calc = stream->runtime->buffer_size - + (stream->runtime->app_pointer - stream->runtime->hw_pointer); + pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc, + stream->runtime->app_pointer, + stream->runtime->hw_pointer); + if (avail_calc >= stream->runtime->buffer_size) + avail_calc -= stream->runtime->buffer_size; + pr_debug("ret avail as %ld\n", avail_calc); + avail->avail = avail_calc; + return avail_calc; +} + +static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) +{ + struct snd_compr_avail avail; + + return snd_compr_calc_avail(stream, &avail); +} + +static int +snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_avail ioctl_avail; + size_t avail; + + avail = snd_compr_calc_avail(stream, &ioctl_avail); + ioctl_avail.avail = avail; + + if (copy_to_user((__u64 __user *)arg, + &ioctl_avail, sizeof(ioctl_avail))) + return -EFAULT; + return 0; +} + +static int snd_compr_write_data(struct snd_compr_stream *stream, + const char __user *buf, size_t count) +{ + void *dstn; + size_t copy; + struct snd_compr_runtime *runtime = stream->runtime; + + dstn = runtime->buffer + runtime->app_pointer; + pr_debug("copying %ld at %lld\n", + (unsigned long)count, runtime->app_pointer); + if (count < runtime->buffer_size - runtime->app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + runtime->app_pointer += count; + } else { + copy = runtime->buffer_size - runtime->app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(runtime->buffer, buf + copy, count - copy)) + return -EFAULT; + runtime->app_pointer = count - copy; + } + /* if DSP cares, let it know data has been written */ + if (stream->ops->ack) + stream->ops->ack(stream, count); + return count; +} + +static ssize_t snd_compr_write(struct file *f, const char __user *buf, + size_t count, loff_t *offset) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval; + + if (snd_BUG_ON(!data)) + return -EFAULT; + + stream = &data->stream; + mutex_lock(&stream->device->lock); + /* write is allowed when stream is running or has been steup */ + if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && + stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + mutex_unlock(&stream->device->lock); + return -EBADFD; + } + + avail = snd_compr_get_avail(stream); + pr_debug("avail returned %ld\n", (unsigned long)avail); + /* calculate how much we can write to buffer */ + if (avail > count) + avail = count; + + if (stream->ops->copy) + retval = stream->ops->copy(stream, buf, avail); + else + retval = snd_compr_write_data(stream, buf, avail); + if (retval > 0) + stream->runtime->total_bytes_available += retval; + + /* while initiating the stream, write should be called before START + * call, so in setup move state */ + if (stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + pr_debug("stream prepared, Houston we are good to go\n"); + } + + mutex_unlock(&stream->device->lock); + return retval; +} + + +static ssize_t snd_compr_read(struct file *f, char __user *buf, + size_t count, loff_t *offset) +{ + return -ENXIO; +} + +static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) +{ + return -ENXIO; +} + +static inline int snd_compr_get_poll(struct snd_compr_stream *stream) +{ + if (stream->direction == SND_COMPRESS_PLAYBACK) + return POLLOUT | POLLWRNORM; + else + return POLLIN | POLLRDNORM; +} + +static unsigned int snd_compr_poll(struct file *f, poll_table *wait) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval = 0; + + if (snd_BUG_ON(!data)) + return -EFAULT; + stream = &data->stream; + if (snd_BUG_ON(!stream)) + return -EFAULT; + + mutex_lock(&stream->device->lock); + if (stream->runtime->state == SNDRV_PCM_STATE_PAUSED || + stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + retval = -EBADFD; + goto out; + } + poll_wait(f, &stream->runtime->sleep, wait); + + avail = snd_compr_get_avail(stream); + pr_debug("avail is %ld\n", (unsigned long)avail); + /* check if we have at least one fragment to fill */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_DRAINING: + /* stream has been woken up after drain is complete + * draining done so set stream state to stopped + */ + retval = snd_compr_get_poll(stream); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + break; + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + if (avail >= stream->runtime->fragment_size) + retval = snd_compr_get_poll(stream); + break; + default: + if (stream->direction == SND_COMPRESS_PLAYBACK) + retval = POLLOUT | POLLWRNORM | POLLERR; + else + retval = POLLIN | POLLRDNORM | POLLERR; + break; + } +out: + mutex_unlock(&stream->device->lock); + return retval; +} + +static int +snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_caps caps; + + if (!stream->ops->get_caps) + return -ENXIO; + + retval = stream->ops->get_caps(stream, &caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, &caps, sizeof(caps))) + retval = -EFAULT; +out: + return retval; +} + +static int +snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_codec_caps *caps; + + if (!stream->ops->get_codec_caps) + return -ENXIO; + + caps = kmalloc(sizeof(*caps), GFP_KERNEL); + if (!caps) + return -ENOMEM; + + retval = stream->ops->get_codec_caps(stream, caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, caps, sizeof(*caps))) + retval = -EFAULT; + +out: + kfree(caps); + return retval; +} + +/* revisit this with snd_pcm_preallocate_xxx */ +static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + unsigned int buffer_size; + void *buffer; + + buffer_size = params->buffer.fragment_size * params->buffer.fragments; + if (stream->ops->copy) { + buffer = NULL; + /* if copy is defined the driver will be required to copy + * the data from core + */ + } else { + buffer = kmalloc(buffer_size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + } + stream->runtime->fragment_size = params->buffer.fragment_size; + stream->runtime->fragments = params->buffer.fragments; + stream->runtime->buffer = buffer; + stream->runtime->buffer_size = buffer_size; + return 0; +} + +static int +snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_params *params; + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) + return -EFAULT; + retval = snd_compr_allocate_buffer(stream, params); + if (retval) { + kfree(params); + return -ENOMEM; + } + retval = stream->ops->set_params(stream, params); + if (retval) + goto out; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + } else + return -EPERM; +out: + kfree(params); + return retval; +} + +static int +snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->get_params) + return -EBADFD; + + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + retval = stream->ops->get_params(stream, params); + if (retval) + goto out; + if (copy_to_user((char __user *)arg, params, sizeof(*params))) + retval = -EFAULT; + +out: + kfree(params); + return retval; +} + +static inline int +snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_tstamp tstamp; + + snd_compr_update_tstamp(stream, &tstamp); + return copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; +} + +static int snd_compr_pause(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_resume(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_start(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_stop(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_drain(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + int retval = -ENOTTY; + + if (snd_BUG_ON(!data)) + return -EFAULT; + stream = &data->stream; + if (snd_BUG_ON(!stream)) + return -EFAULT; + mutex_lock(&stream->device->lock); + switch (_IOC_NR(cmd)) { + case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): + put_user(SNDRV_COMPRESS_VERSION, + (int __user *)arg) ? -EFAULT : 0; + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): + retval = snd_compr_get_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS): + retval = snd_compr_get_codec_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): + retval = snd_compr_set_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): + retval = snd_compr_get_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_TSTAMP): + retval = snd_compr_tstamp(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_AVAIL): + retval = snd_compr_ioctl_avail(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_PAUSE): + retval = snd_compr_pause(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_RESUME): + retval = snd_compr_resume(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_START): + retval = snd_compr_start(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_STOP): + retval = snd_compr_stop(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_DRAIN): + retval = snd_compr_drain(stream); + break; + } + mutex_unlock(&stream->device->lock); + return retval; +} + +static const struct file_operations snd_compr_file_ops = { + .owner = THIS_MODULE, + .open = snd_compr_open, + .release = snd_compr_free, + .write = snd_compr_write, + .read = snd_compr_read, + .unlocked_ioctl = snd_compr_ioctl, + .mmap = snd_compr_mmap, + .poll = snd_compr_poll, +}; + +static int snd_compress_dev_register(struct snd_device *device) +{ + int ret = -EINVAL; + char str[16]; + struct snd_compr *compr; + + if (snd_BUG_ON(!device || !device->device_data)) + return -EBADFD; + compr = device->device_data; + + sprintf(str, "comprC%iD%i", compr->card->number, compr->device); + pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + compr->direction); + /* register compressed device */ + ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS, compr->card, + compr->device, &snd_compr_file_ops, compr, str); + if (ret < 0) { + pr_err("snd_register_device failed\n %d", ret); + return ret; + } + return ret; + +} + +static int snd_compress_dev_disconnect(struct snd_device *device) +{ + struct snd_compr *compr; + + compr = device->device_data; + snd_unregister_device(compr->direction, compr->card, compr->device); + return 0; +} + +/* + * snd_compress_new: create new compress device + * @card: sound card pointer + * @device: device number + * @dirn: device direction, should be of type enum snd_compr_direction + * @compr: compress device pointer + */ +int snd_compress_new(struct snd_card *card, int device, + int dirn, struct snd_compr *compr) +{ + static struct snd_device_ops ops = { + .dev_free = NULL, + .dev_register = snd_compress_dev_register, + .dev_disconnect = snd_compress_dev_disconnect, + }; + + compr->card = card; + compr->device = device; + compr->direction = dirn; + return snd_device_new(card, SNDRV_DEV_COMPRESS, compr, &ops); +} +EXPORT_SYMBOL_GPL(snd_compress_new); + +static int snd_compress_add_device(struct snd_compr *device) +{ + int ret; + + if (!device->card) + return -EINVAL; + + /* register the card */ + ret = snd_card_register(device->card); + if (ret) + goto out; + return 0; + +out: + pr_err("failed with %d\n", ret); + return ret; + +} + +static int snd_compress_remove_device(struct snd_compr *device) +{ + return snd_card_free(device->card); +} + +/** + * snd_compress_register - register compressed device + * + * @device: compressed device to register + */ +int snd_compress_register(struct snd_compr *device) +{ + int retval; + + if (device->name == NULL || device->dev == NULL || device->ops == NULL) + return -EINVAL; + + pr_debug("Registering compressed device %s\n", device->name); + if (snd_BUG_ON(!device->ops->open)) + return -EINVAL; + if (snd_BUG_ON(!device->ops->free)) + return -EINVAL; + if (snd_BUG_ON(!device->ops->set_params)) + return -EINVAL; + if (snd_BUG_ON(!device->ops->trigger)) + return -EINVAL; + + mutex_init(&device->lock); + + /* register a compressed card */ + mutex_lock(&device_mutex); + retval = snd_compress_add_device(device); + mutex_unlock(&device_mutex); + return retval; +} +EXPORT_SYMBOL_GPL(snd_compress_register); + +int snd_compress_deregister(struct snd_compr *device) +{ + pr_debug("Removing compressed device %s\n", device->name); + mutex_lock(&device_mutex); + snd_compress_remove_device(device); + mutex_unlock(&device_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_compress_deregister); + +static int __init snd_compress_init(void) +{ + return 0; +} + +static void __exit snd_compress_exit(void) +{ +} + +module_init(snd_compress_init); +module_exit(snd_compress_exit); + +MODULE_DESCRIPTION("ALSA Compressed offload framework"); +MODULE_AUTHOR("Vinod Koul vinod.koul@linux.intel.com"); +MODULE_LICENSE("GPL v2");

From: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com

The patch adds the documentation file explaining the API

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Signed-off-by: Vinod Koul vinod.koul@linux.intel.com --- Documentation/sound/alsa/snd_compress_data.txt | 184 ++++++++++++++++++++++++ 1 files changed, 184 insertions(+), 0 deletions(-) create mode 100644 Documentation/sound/alsa/snd_compress_data.txt

diff --git a/Documentation/sound/alsa/snd_compress_data.txt b/Documentation/sound/alsa/snd_compress_data.txt new file mode 100644 index 0000000..4b0d0fc --- /dev/null +++ b/Documentation/sound/alsa/snd_compress_data.txt @@ -0,0 +1,184 @@ + 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_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. + +- 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; + 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 +- 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.

The patch adds the various definations used to define the encoder and decoder parameters

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com Reviewed-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/snd_compress_params.h | 395 +++++++++++++++++++++++++++++++++++ 1 files changed, 395 insertions(+), 0 deletions(-) create mode 100644 include/sound/snd_compress_params.h

diff --git a/include/sound/snd_compress_params.h b/include/sound/snd_compress_params.h new file mode 100644 index 0000000..79b681d --- /dev/null +++ b/include/sound/snd_compress_params.h @@ -0,0 +1,395 @@ +/* + * 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. + * + */ +#ifndef __SND_COMPRESS_PARAMS_H +#define __SND_COMPRESS_PARAMS_H + +/* AUDIO CODECS SUPPORTED */ +#define MAX_NUM_CODECS 32 +#define MAX_NUM_CODEC_DESCRIPTORS 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) +#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) +#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) + +/* + * 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) + +#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002) +#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002) + +/* <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 snd_enc_wma { + __u32 super_block_align; /* WMA Type-specific data */ +}; + + +/** + * struct snd_enc_vorbis + * @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. + * @max_bit_rate: Enabled only if managed is TRUE + * @min_bit_rate: Enabled only if 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 snd_enc_vorbis { + int quality; + __u32 managed; + __u32 max_bit_rate; + __u32 min_bit_rate; + __u32 downmix; +}; + + +/** + * struct snd_enc_real + * @quant_bits: number of coupling quantization bits in the stream + * @start_region: coupling start region in the stream + * @num_regions: number of regions value + * + * These options were extracted from the OpenMAX IL spec + */ + +struct snd_enc_real { + __u32 quant_bits; + __u32 start_region; + __u32 num_regions; +}; + +/** + * struct snd_enc_flac + * @num: serial number, valid only for OGG formats + * needs to be set by application + * @gain: Add replay gain 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 needed. + * + * 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 snd_enc_flac { + __u32 num; + __u32 gain; +}; + +struct snd_enc_generic { + __u32 bw; /* encoder bandwidth */ + int reserved[15]; +}; + +union snd_codec_options { + struct snd_enc_wma wma; + struct snd_enc_vorbis vorbis; + struct snd_enc_real real; + struct snd_enc_flac flac; + struct snd_enc_generic generic; +}; + +/** struct snd_codec_desc - description of codec capabilities + * @max_ch: Maximum number of audio channels + * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this + * @bit_rate: Indexed array containing supported bit rates + * @num_bitrates: Number of valid values in bit_rate array + * @rate_control: value is specified by SND_RATECONTROLMODE defines. + * @profiles: Supported profiles. See SND_AUDIOPROFILE defines. + * @modes: Supported modes. See SND_AUDIOMODE defines + * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines + * @reserved: reserved for future use + * + * This structure provides a scalar value for profiles, modes and stream + * format fields. + * If an implementation supports multiple combinations, they will be listed as + * codecs with different descriptors, for example there would be 2 descriptors + * for AAC-RAW and AAC-ADTS. + * This entails some redundancy but makes it easier to avoid invalid + * configurations. + * + */ + +struct snd_codec_desc { + __u32 max_ch; + __u32 sample_rates; + __u32 bit_rate[MAX_NUM_BITRATES]; + __u32 num_bitrates; + __u32 rate_control; + __u32 profiles; + __u32 modes; + __u32 formats; + __u32 reserved[16]; +}; + +/** struct snd_codec + * @id: Identifies the supported audio encoder/decoder. + * See SND_AUDIOCODEC macros. + * @ch_in: Number of input audio channels + * @ch_out: Number of output channels. In case of contradiction between + * this field and the channelMode field, the channelMode field + * overrides. + * @sample_rate: Audio sample rate of input data + * @bit_rate: Bitrate of encoded data. May be ignored by decoders + * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines. + * Encoders may rely on profiles for quality levels. + * May be ignored by decoders. + * @profile: Mandatory for encoders, can be mandatory for specific + * decoders as well. See SND_AUDIOPROFILE defines. + * @level: Supported level (Only used by WMA at the moment) + * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines + * @format: Format of encoded bistream. Mandatory when defined. + * See SND_AUDIOSTREAMFORMAT defines. + * @align: 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 snd_codec { + __u32 id; + __u32 ch_in; + __u32 ch_out; + __u32 sample_rate; + __u32 bit_rate; + __u32 rate_control; + __u32 profile; + __u32 level; + __u32 ch_mode; + __u32 format; + __u32 align; + union snd_codec_options options; + __u32 reserved[3]; +}; + +#endif

This patch adds core.c, the file which implements the ioctls and registers the devices

Signed-off-by: Vinod Koul vinod.koul@linux.intel.com Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/core/compress_offload.c | 701 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 701 insertions(+), 0 deletions(-) create mode 100644 sound/core/compress_offload.c

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c new file mode 100644 index 0000000..b77f8fa --- /dev/null +++ b/sound/core/compress_offload.c @@ -0,0 +1,701 @@ +/* + * core.c - compress offload core + * + * Copyright (C) 2011 Intel Corporation + * Authors: Vinod Koul vinod.koul@linux.intel.com + * Pierre-Louis Bossart pierre-louis.bossart@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#define FORMAT(fmt) "%s: %d: " fmt, __func__, __LINE__ +#define pr_fmt(fmt) KBUILD_MODNAME ": " FORMAT(fmt) + +#include <linux/file.h> +#include <linux/fs.h> +#include <linux/list.h> +#include <linux/mm.h> +#include <linux/mutex.h> +#include <linux/poll.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <linux/uio.h> +#include <linux/uaccess.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/snd_compress_params.h> +#include <sound/compress_offload.h> +#include <sound/compress_driver.h> + +/* TODO: + * - add substream support for multiple devices in case of + * SND_DYNAMIC_MINORS is not used + * - Multiple node representation + * driver should be able to register multiple nodes + */ + +static DEFINE_MUTEX(device_mutex); + +struct snd_compr_file { + unsigned long caps; + struct snd_compr_stream stream; +}; + +static int snd_compr_open(struct inode *inode, struct file *f) +{ + struct snd_compr *compr; + struct snd_compr_file *data; + struct snd_compr_runtime *runtime; + int ret, type; + int maj = imajor(inode); + + if (f->f_flags & O_WRONLY) + type = SNDRV_DEVICE_TYPE_COMPR_PLAYBACK; + else if (f->f_flags & O_RDONLY) + type = SNDRV_DEVICE_TYPE_COMPR_CAPTURE; + else { + pr_err("invalid direction\n"); + return -EINVAL; + } + + if (maj == snd_major) + compr = snd_lookup_minor_data(iminor(inode), type); + else + return -ENXIO; + + if (compr == NULL) { + pr_err("no device data!!!\n"); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + data->stream.ops = compr->ops; + data->stream.direction = type; + data->stream.private_data = compr->private_data; + data->stream.device = compr; + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) { + kfree(data); + return -ENOMEM; + } + runtime->state = SNDRV_PCM_STATE_OPEN; + init_waitqueue_head(&runtime->sleep); + data->stream.runtime = runtime; + f->private_data = (void *)data; + mutex_lock(&compr->lock); + ret = compr->ops->open(&data->stream); + mutex_unlock(&compr->lock); + if (ret) { + kfree(runtime); + kfree(data); + } + return ret; +} + +static int snd_compr_free(struct inode *inode, struct file *f) +{ + struct snd_compr_file *data = f->private_data; + data->stream.ops->free(&data->stream); + kfree(data->stream.runtime->buffer); + kfree(data->stream.runtime); + kfree(data); + return 0; +} + +static void snd_compr_update_tstamp(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + if (!stream->ops->pointer) + return; + stream->ops->pointer(stream, tstamp); + pr_debug("dsp consumed till %d total %d bytes\n", + tstamp->copied_bytes, tstamp->copied_total); + stream->runtime->hw_pointer = tstamp->copied_bytes; + stream->runtime->bytes_copied = tstamp->copied_total; +} + +static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, + struct snd_compr_avail *avail) +{ + long avail_calc; /*this needs to be signed variable */ + + snd_compr_update_tstamp(stream, &avail->tstamp); + + if (stream->runtime->bytes_written == 0 && + stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + pr_debug("detected init and someone forgot to do a write\n"); + return stream->runtime->buffer_size; + } + pr_debug("app wrote %d, DSP consumed %d\n", + stream->runtime->bytes_written, stream->runtime->bytes_copied); + if (stream->runtime->bytes_written == stream->runtime->bytes_copied) { + pr_debug("both pointers are same, returning full avail\n"); + return stream->runtime->buffer_size; + } + + avail_calc = stream->runtime->buffer_size - + (stream->runtime->app_pointer - stream->runtime->hw_pointer); + pr_debug("calc avail as %ld, app_ptr %d, hw+ptr %d\n", avail_calc, + stream->runtime->app_pointer, stream->runtime->hw_pointer); + if (avail_calc >= stream->runtime->buffer_size) + avail_calc -= stream->runtime->buffer_size; + pr_debug("ret avail as %ld\n", avail_calc); + avail->avail = avail_calc; + return avail_calc; +} + +static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) +{ + struct snd_compr_avail avail; + + return snd_compr_calc_avail(stream, &avail); +} + +static int +snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_avail ioctl_avail; + size_t avail; + + avail = snd_compr_calc_avail(stream, &ioctl_avail); + ioctl_avail.avail = avail; + + if (copy_to_user((unsigned long __user *)arg, + &ioctl_avail, sizeof(ioctl_avail))) + return -EFAULT; + return 0; +} + +static int snd_compr_write_data(struct snd_compr_stream *stream, + const char __user *buf, size_t count) +{ + void *dstn; + size_t copy; + struct snd_compr_runtime *runtime = stream->runtime; + + dstn = runtime->buffer + runtime->app_pointer; + pr_debug("copying %d at %d\n", count, runtime->app_pointer); + if (count < runtime->buffer_size - runtime->app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + runtime->app_pointer += count; + } else { + copy = runtime->buffer_size - runtime->app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(runtime->buffer, buf + copy, count - copy)) + return -EFAULT; + runtime->app_pointer = count - copy; + } + /* if DSP cares, let it know data has been written */ + if (stream->ops->ack) + stream->ops->ack(stream, count); + return count; +} + +static ssize_t snd_compr_write(struct file *f, const char __user *buf, + size_t count, loff_t *offset) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval; + + BUG_ON(!data); + stream = &data->stream; + mutex_lock(&stream->device->lock); + /* write is allowed when stream is running or has been steup */ + if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && + stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + mutex_unlock(&stream->device->lock); + return -EPERM; + } + + avail = snd_compr_get_avail(stream); + pr_debug("avail returned %d\n", avail); + /* calculate how much we can write to buffer */ + if (avail > count) + avail = count; + + if (stream->ops->copy) + retval = stream->ops->copy(stream, buf, avail); + else + retval = snd_compr_write_data(stream, buf, avail); + if (retval > 0) + stream->runtime->bytes_written += retval; + + /* while initiating the stream, write should be called before START + * call, so in setup move state */ + if (stream->runtime->state == SNDRV_PCM_STATE_SETUP) { + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + pr_debug("stream prepared, Houston we are good to go\n"); + } + + mutex_unlock(&stream->device->lock); + return retval; +} + + +static ssize_t snd_compr_read(struct file *f, char __user *buf, + size_t count, loff_t *offset) +{ + return -ENXIO; +} + +static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) +{ + return -ENXIO; +} + +static unsigned int snd_compr_poll(struct file *f, poll_table *wait) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + size_t avail; + int retval = 0; + + BUG_ON(!data); + stream = &data->stream; + + mutex_lock(&stream->device->lock); + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + retval = -ENXIO; + goto out; + } + poll_wait(f, &stream->runtime->sleep, wait); + + avail = snd_compr_get_avail(stream); + pr_debug("avail is %d\n", avail); + /* check if we have at least one fragment to fill */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + if (avail >= stream->runtime->fragment_size) { + if (stream->direction == SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + retval = POLLOUT | POLLWRNORM; + else + retval = POLLIN | POLLRDNORM; + break; + } + /* Fall through */ + case SNDRV_PCM_STATE_DRAINING: + break; + default: + if (stream->direction == SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + retval = POLLOUT | POLLWRNORM | POLLERR; + else + retval = POLLIN | POLLRDNORM | POLLERR; + break; + } +out: + mutex_unlock(&stream->device->lock); + return retval; +} + +static int +snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_caps caps; + + if (!stream->ops->get_caps) + return -ENXIO; + + retval = stream->ops->get_caps(stream, &caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, &caps, sizeof(caps))) + retval = -EFAULT; +out: + return retval; +} + +static int +snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) +{ + int retval; + struct snd_compr_codec_caps *caps; + + if (!stream->ops->get_codec_caps) + return -ENXIO; + + caps = kmalloc(sizeof(*caps), GFP_KERNEL); + if (!caps) + return -ENOMEM; + + retval = stream->ops->get_codec_caps(stream, caps); + if (retval) + goto out; + if (copy_to_user((void __user *)arg, caps, sizeof(*caps))) + retval = -EFAULT; + +out: + kfree(caps); + return retval; +} + +/* revisit this with snd_pcm_preallocate_xxx */ +static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + unsigned int buffer_size; + void *buffer; + + buffer_size = params->buffer.fragment_size * params->buffer.fragments; + if (stream->ops->copy) { + buffer = NULL; + /* if copy is defined the driver will be required to copy + * the data from core + */ + } else { + buffer = kmalloc(buffer_size, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + } + stream->runtime->fragment_size = params->buffer.fragment_size; + stream->runtime->fragments = params->buffer.fragments; + stream->runtime->buffer = buffer; + stream->runtime->buffer_size = buffer_size; + return 0; +} + +static int +snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_params *params; + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) + return -EFAULT; + retval = snd_compr_allocate_buffer(stream, params); + if (retval) { + kfree(params); + return -ENOMEM; + } + retval = stream->ops->set_params(stream, params); + if (retval) + goto out; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + } else + return -EPERM; +out: + kfree(params); + return retval; +} + +static int +snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->get_params) + return -ENXIO; + + params = kmalloc(sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + retval = stream->ops->get_params(stream, params); + if (retval) + goto out; + if (copy_to_user((char __user *)arg, params, sizeof(*params))) + retval = -EFAULT; + +out: + kfree(params); + return retval; +} + +static inline int +snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_tstamp tstamp; + + snd_compr_update_tstamp(stream, &tstamp); + return copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; +} + +static int snd_compr_pause(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_resume(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_start(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + return retval; +} + +static int snd_compr_stop(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static int snd_compr_drain(struct snd_compr_stream *stream) +{ + int retval; + + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); + if (!retval) { + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + wake_up(&stream->runtime->sleep); + } + return retval; +} + +static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) +{ + struct snd_compr_file *data = f->private_data; + struct snd_compr_stream *stream; + int retval = -ENOTTY; + + BUG_ON(!data); + stream = &data->stream; + BUG_ON(!stream); + mutex_lock(&stream->device->lock); + switch (_IOC_NR(cmd)) { + case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): + put_user(SNDRV_COMPRESS_VERSION, + (int __user *)arg) ? -EFAULT : 0; + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): + retval = snd_compr_get_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS): + retval = snd_compr_get_codec_caps(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): + retval = snd_compr_set_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): + retval = snd_compr_get_params(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_TSTAMP): + retval = snd_compr_tstamp(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_AVAIL): + retval = snd_compr_ioctl_avail(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_PAUSE): + retval = snd_compr_pause(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_RESUME): + retval = snd_compr_resume(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_START): + retval = snd_compr_start(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_STOP): + retval = snd_compr_stop(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_DRAIN): + retval = snd_compr_drain(stream); + break; + } + mutex_unlock(&stream->device->lock); + return retval; +} + +static const struct file_operations snd_compr_file_ops = { + .owner = THIS_MODULE, + .open = snd_compr_open, + .release = snd_compr_free, + .write = snd_compr_write, + .read = snd_compr_read, + .unlocked_ioctl = snd_compr_ioctl, + .mmap = snd_compr_mmap, + .poll = snd_compr_poll, +}; + +static int snd_compress_dev_register(struct snd_device *device) +{ + int ret = -EINVAL; + char str[16]; + struct snd_compr *compr; + + if (snd_BUG_ON(!device || !device->device_data)) + return -ENXIO; + compr = device->device_data; + + sprintf(str, "comprC%iD%i", compr->card->number, compr->device); + pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + compr->direction); + /* register compressed device */ + ret = snd_register_device(compr->direction, compr->card, compr->device, + &snd_compr_file_ops, compr, str); + if (ret < 0) { + pr_err("snd_register_device failed\n %d", ret); + return ret; + } + return ret; + +} + +static int snd_compress_dev_disconnect(struct snd_device *device) +{ + struct snd_compr *compr; + + compr = device->device_data; + snd_unregister_device(compr->direction, compr->card, compr->device); + return 0; +} + +int snd_compress_new(struct snd_card *card, int device, + int type, struct snd_compr *compr) +{ + static struct snd_device_ops ops = { + .dev_free = NULL, + .dev_register = snd_compress_dev_register, + .dev_disconnect = snd_compress_dev_disconnect, + }; + + compr->card = card; + compr->device = device; + compr->direction = type; + return snd_device_new(card, SNDRV_DEV_COMPRESS, compr, &ops); +} +EXPORT_SYMBOL_GPL(snd_compress_new); + +static int snd_compress_add_device(struct snd_compr *device) +{ + int ret; + + if (!device->card) + return -EINVAL; + + /* register the card */ + ret = snd_card_register(device->card); + if (ret) + goto out; + return 0; + +out: + pr_err("failed with %d\n", ret); + return ret; + +} + +static int snd_compress_remove_device(struct snd_compr *device) +{ + return snd_card_free(device->card); +} + +/** + * snd_compress_register - register compressed device + * + * @device: compressed device to register + */ +int snd_compress_register(struct snd_compr *device) +{ + int retval; + + if (device->name == NULL || device->dev == NULL || device->ops == NULL) + return -EINVAL; + + pr_debug("Registering compressed device %s\n", device->name); + BUG_ON(!device->ops->open); + BUG_ON(!device->ops->free); + BUG_ON(!device->ops->set_params); + BUG_ON(!device->ops->trigger); + + mutex_init(&device->lock); + + /* register a compressed card */ + mutex_lock(&device_mutex); + retval = snd_compress_add_device(device); + mutex_unlock(&device_mutex); + return retval; +} +EXPORT_SYMBOL_GPL(snd_compress_register); + +int snd_compress_deregister(struct snd_compr *device) +{ + pr_debug("Removing compressed device %s\n", device->name); + mutex_lock(&device_mutex); + snd_compress_remove_device(device); + mutex_unlock(&device_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_compress_deregister); + +static int __init snd_compress_init(void) +{ + return 0; +} + +static void __exit snd_compress_exit(void) +{ +} + +module_init(snd_compress_init); +module_exit(snd_compress_exit); + +MODULE_DESCRIPTION("ALSA Compressed offload framework"); +MODULE_AUTHOR("Vinod Koul vinod.koul@linux.intel.com"); +MODULE_LICENSE("GPL v2");