From 418255d9d977cf3d0a3bde8e441af1375caef9db Mon Sep 17 00:00:00 2001 From: Matt Wagantall Date: Thu, 11 Sep 2014 20:43:41 -0700 Subject: [PATCH] ASoC: msm: Add compress audio playback code base Support bits and pieces to make msm-compress-q6-v2.c usable will be added in subsequent commits. Signed-off-by: Matt Wagantall --- sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c | 950 +++++++++++++++++++++ 1 file changed, 950 insertions(+) create mode 100644 sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c new file mode 100644 index 000000000000..a66a8e92ab84 --- /dev/null +++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c @@ -0,0 +1,950 @@ +/* Copyright (c) 2012-2013, The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * 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. + */ + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#include +#include + +#include "msm-pcm-routing-v2.h" +#include "audio_ocmem.h" + +/* Default values used if user space does not set */ +#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024) +#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) +#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) +#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) + +#define COMPRESSED_LR_VOL_MAX_STEPS 0x2000 +const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0, + COMPRESSED_LR_VOL_MAX_STEPS); + +struct msm_compr_pdata { + atomic_t audio_ocmem_req; + struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; + uint32_t volume[MSM_FRONTEND_DAI_MAX][2]; /* For both L & R */ +}; + +struct msm_compr_audio { + struct snd_compr_stream *cstream; + struct snd_compr_caps compr_cap; + struct snd_compr_codec_caps codec_caps; + struct snd_compr_params codec_param; + struct audio_client *audio_client; + + uint32_t codec; + void *buffer; /* virtual address */ + uint32_t buffer_paddr; /* physical address */ + uint32_t app_pointer; + uint32_t buffer_size; + uint32_t byte_offset; + uint32_t copied_total; + uint32_t bytes_received; + + uint16_t session_id; + + uint32_t sample_rate; + uint32_t num_channels; + + uint32_t cmd_ack; + uint32_t cmd_interrupt; + uint32_t drain_ready; + + atomic_t start; + atomic_t eos; + atomic_t drain; + + wait_queue_head_t eos_wait; + wait_queue_head_t drain_wait; + wait_queue_head_t flush_wait; + + spinlock_t lock; +}; + +static int msm_compr_set_volume(struct snd_compr_stream *cstream, + uint32_t volume_l, uint32_t volume_r) +{ + struct msm_compr_audio *prtd; + int rc = 0; + + pr_debug("%s: volume_l %d volume_r %d\n", + __func__, volume_l, volume_r); + prtd = cstream->runtime->private_data; + if (prtd && prtd->audio_client) { + if (volume_l != volume_r) { + pr_debug("%s: call q6asm_set_lrgain\n", __func__); + rc = q6asm_set_lrgain(prtd->audio_client, + volume_l, volume_r); + } else { + pr_debug("%s: call q6asm_set_volume\n", __func__); + rc = q6asm_set_volume(prtd->audio_client, volume_l); + } + if (rc < 0) { + pr_err("%s: Send Volume command failed rc=%d\n", + __func__, rc); + } + } + + return rc; +} + +static int msm_compr_send_buffer(struct msm_compr_audio *prtd) +{ + int buffer_length; + int bytes_available; + struct audio_aio_write_param param; + + if (!atomic_read(&prtd->start)) { + pr_err("%s: stream is not in started state\n", __func__); + return -EINVAL; + } + pr_debug("%s: bytes_received = %d copied_total = %d\n", + __func__, prtd->bytes_received, prtd->copied_total); + buffer_length = prtd->codec_param.buffer.fragment_size; + bytes_available = prtd->bytes_received - prtd->copied_total; + if (bytes_available < prtd->codec_param.buffer.fragment_size) + buffer_length = bytes_available; + + if (buffer_length == 0) { + pr_debug("Recieved a zero length buffer-break out\n"); + if (atomic_read(&prtd->drain)) { + prtd->drain_ready = 1; + wake_up(&prtd->drain_wait); + atomic_set(&prtd->drain, 0); + } + return 0; + } + + param.paddr = prtd->buffer_paddr + prtd->byte_offset; + param.len = buffer_length; + param.msw_ts = 0; + param.lsw_ts = 0; + param.flags = NO_TIMESTAMP; + param.uid = buffer_length; + param.metadata_len = 0; + + pr_debug("%s: sending %d bytes to DSP byte_offset = %d\n", + __func__, buffer_length, prtd->byte_offset); + if (q6asm_async_write(prtd->audio_client, ¶m) < 0) + pr_err("%s:q6asm_async_write failed\n", __func__); + + return 0; +} + +static void compr_event_handler(uint32_t opcode, + uint32_t token, uint32_t *payload, void *priv) +{ + struct msm_compr_audio *prtd = priv; + struct snd_compr_stream *cstream = prtd->cstream; + uint32_t chan_mode = 0; + uint32_t sample_rate = 0; + + pr_debug("%s opcode =%08x\n", __func__, opcode); + switch (opcode) { + case ASM_DATA_EVENT_WRITE_DONE_V2: + pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n"); + spin_lock_irq(&prtd->lock); + prtd->byte_offset += token; + prtd->copied_total += token; + if (prtd->byte_offset >= prtd->buffer_size) + prtd->byte_offset -= prtd->buffer_size; + + snd_compr_fragment_elapsed(cstream); + if (atomic_read(&prtd->start)) + msm_compr_send_buffer(prtd); + spin_unlock_irq(&prtd->lock); + break; + case ASM_DATA_EVENT_RENDERED_EOS: + pr_debug("ASM_DATA_CMDRSP_EOS\n"); + if (atomic_read(&prtd->eos)) { + pr_debug("ASM_DATA_CMDRSP_EOS wake up\n"); + prtd->cmd_ack = 1; + wake_up(&prtd->eos_wait); + atomic_set(&prtd->eos, 0); + } + break; + case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY: + case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY: { + pr_debug("ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY\n"); + chan_mode = payload[1] >> 16; + sample_rate = payload[2] >> 16; + if (prtd && (chan_mode != prtd->num_channels || + sample_rate != prtd->sample_rate)) { + prtd->num_channels = chan_mode; + prtd->sample_rate = sample_rate; + } + } + case APR_BASIC_RSP_RESULT: { + switch (payload[0]) { + case ASM_SESSION_CMD_RUN_V2: + /* check if the first buffer need to be sent to DSP */ + pr_debug("ASM_SESSION_CMD_RUN_V2\n"); + if (!prtd->copied_total) + msm_compr_send_buffer(prtd); + break; + case ASM_STREAM_CMD_FLUSH: + pr_debug("ASM_STREAM_CMD_FLUSH\n"); + prtd->cmd_ack = 1; + wake_up(&prtd->flush_wait); + break; + default: + break; + } + break; + } + case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3: + pr_debug("ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n"); + break; + default: + pr_debug("Not Supported Event opcode[0x%x]\n", opcode); + break; + } +} + +static void populate_codec_list(struct msm_compr_audio *prtd) +{ + pr_debug("%s\n", __func__); + prtd->compr_cap.direction = SND_COMPRESS_PLAYBACK; + prtd->compr_cap.min_fragment_size = + COMPR_PLAYBACK_MIN_FRAGMENT_SIZE; + prtd->compr_cap.max_fragment_size = + COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; + prtd->compr_cap.min_fragments = + COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; + prtd->compr_cap.max_fragments = + COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + prtd->compr_cap.num_codecs = 2; + prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3; + prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC; +} + +static int msm_compr_configure_dsp(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; + uint16_t bits_per_sample = 16; + int dir = IN, ret = 0; + struct asm_softpause_params softpause = { + .enable = SOFT_PAUSE_ENABLE, + .period = SOFT_PAUSE_PERIOD, + .step = SOFT_PAUSE_STEP, + .rampingcurve = SOFT_PAUSE_CURVE_LINEAR, + }; + struct asm_softvolume_params softvol = { + .period = SOFT_VOLUME_PERIOD, + .step = SOFT_VOLUME_STEP, + .rampingcurve = SOFT_VOLUME_CURVE_LINEAR, + }; + + pr_debug("%s\n", __func__); + ret = q6asm_open_write_v2(prtd->audio_client, + prtd->codec, bits_per_sample); + if (ret < 0) { + pr_err("%s: Session out open failed\n", __func__); + return -ENOMEM; + } + + pr_debug("%s be_id %d\n", __func__, soc_prtd->dai_link->be_id); + msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id, + prtd->audio_client->perf_mode, + prtd->session_id, + SNDRV_PCM_STREAM_PLAYBACK); + + ret = msm_compr_set_volume(cstream, 0, 0); + if (ret < 0) + pr_err("%s : Set Volume failed : %d", __func__, ret); + + ret = q6asm_set_softpause(prtd->audio_client, + &softpause); + if (ret < 0) + pr_err("%s: Send SoftPause Param failed ret=%d\n", + __func__, ret); + + ret = q6asm_set_softvolume(prtd->audio_client, &softvol); + if (ret < 0) + pr_err("%s: Send SoftVolume Param failed ret=%d\n", + __func__, ret); + + ret = q6asm_set_io_mode(prtd->audio_client, + (COMPRESSED_IO | ASYNC_IO_MODE)); + if (ret < 0) { + pr_err("%s: Set IO mode failed\n", __func__); + return -EINVAL; + } + + runtime->fragments = prtd->codec_param.buffer.fragments; + runtime->fragment_size = prtd->codec_param.buffer.fragment_size; + pr_debug("allocate %d buffers each of size %d\n", + runtime->fragments, + runtime->fragment_size); + ret = q6asm_audio_client_buf_alloc_contiguous(dir, + prtd->audio_client, + runtime->fragment_size, + runtime->fragments); + if (ret < 0) { + pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); + return -ENOMEM; + } + + prtd->byte_offset = 0; + prtd->copied_total = 0; + prtd->app_pointer = 0; + prtd->bytes_received = 0; + prtd->buffer = prtd->audio_client->port[dir].buf[0].data; + prtd->buffer_paddr = prtd->audio_client->port[dir].buf[0].phys; + prtd->buffer_size = runtime->fragments * runtime->fragment_size; + + return 0; +} + +static int msm_compr_open(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct msm_compr_audio *prtd; + struct msm_compr_pdata *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + + pr_debug("%s\n", __func__); + prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); + if (prtd == NULL) { + pr_err("Failed to allocate memory for msm_compr_audio\n"); + return -ENOMEM; + } + + prtd->cstream = cstream; + pdata->cstream[rtd->dai_link->be_id] = cstream; + prtd->audio_client = q6asm_audio_client_alloc( + (app_cb)compr_event_handler, prtd); + if (!prtd->audio_client) { + pr_err("%s: Could not allocate memory\n", __func__); + kfree(prtd); + return -ENOMEM; + } + + pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); + prtd->audio_client->perf_mode = false; + prtd->session_id = prtd->audio_client->session; + prtd->codec = FORMAT_MP3; + prtd->bytes_received = 0; + prtd->copied_total = 0; + prtd->byte_offset = 0; + prtd->sample_rate = 44100; + prtd->num_channels = 2; + prtd->drain_ready = 0; + + spin_lock_init(&prtd->lock); + + atomic_set(&prtd->eos, 0); + atomic_set(&prtd->start, 0); + atomic_set(&prtd->drain, 0); + + init_waitqueue_head(&prtd->eos_wait); + init_waitqueue_head(&prtd->drain_wait); + init_waitqueue_head(&prtd->flush_wait); + + runtime->private_data = prtd; + populate_codec_list(prtd); + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + if (!atomic_cmpxchg(&pdata->audio_ocmem_req, 0, 1)) + audio_ocmem_process_req(AUDIO, true); + else + atomic_inc(&pdata->audio_ocmem_req); + pr_debug("%s: ocmem_req: %d\n", __func__, + atomic_read(&pdata->audio_ocmem_req)); + } else { + pr_err("%s: Unsupported stream type", __func__); + } + + return 0; +} + +static int msm_compr_free(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; + struct msm_compr_pdata *pdata = + snd_soc_platform_get_drvdata(soc_prtd->platform); + int dir = IN, ret = 0; + + pr_debug("%s\n", __func__); + pdata->cstream[soc_prtd->dai_link->be_id] = NULL; + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + if (atomic_read(&pdata->audio_ocmem_req) > 1) + atomic_dec(&pdata->audio_ocmem_req); + else if (atomic_cmpxchg(&pdata->audio_ocmem_req, 1, 0)) + audio_ocmem_process_req(AUDIO, false); + + msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id, + SNDRV_PCM_STREAM_PLAYBACK); + } + + pr_debug("%s: ocmem_req: %d\n", __func__, + atomic_read(&pdata->audio_ocmem_req)); + + ret = wait_event_timeout(prtd->eos_wait, + prtd->cmd_ack, 5 * HZ); + if (!ret) + pr_err("%s: CMD_EOS failed\n", __func__); + + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + + q6asm_audio_client_buf_free_contiguous(dir, + prtd->audio_client); + + q6asm_audio_client_free(prtd->audio_client); + + kfree(prtd); + + return 0; +} + +/* compress stream operations */ +static int msm_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + int ret = 0; + + pr_debug("%s\n", __func__); + + memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params)); + + /* ToDo: remove duplicates */ + prtd->num_channels = prtd->codec_param.codec.ch_in; + + switch (prtd->codec_param.codec.sample_rate) { + case SNDRV_PCM_RATE_8000: + prtd->sample_rate = 8000; + break; + case SNDRV_PCM_RATE_11025: + prtd->sample_rate = 11025; + break; + /* ToDo: What about 12K and 24K sample rates ? */ + case SNDRV_PCM_RATE_16000: + prtd->sample_rate = 16000; + break; + case SNDRV_PCM_RATE_22050: + prtd->sample_rate = 22050; + break; + case SNDRV_PCM_RATE_32000: + prtd->sample_rate = 32000; + break; + case SNDRV_PCM_RATE_44100: + prtd->sample_rate = 44100; + break; + case SNDRV_PCM_RATE_48000: + prtd->sample_rate = 48000; + break; + } + + pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate); + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + pr_debug("SND_AUDIOCODEC_MP3\n"); + prtd->codec = FORMAT_MP3; + break; + } + + case SND_AUDIOCODEC_AAC: { + pr_debug("SND_AUDIOCODEC_AAC\n"); + prtd->codec = FORMAT_MPEG4_AAC; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + ret = msm_compr_configure_dsp(cstream); + + return ret; +} + +static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct msm_compr_pdata *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + uint32_t *volume = pdata->volume[rtd->dai_link->be_id]; + int rc = 0; + + if (cstream->direction != SND_COMPRESS_PLAYBACK) { + pr_err("%s: Unsupported stream type\n", __func__); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__); + atomic_set(&prtd->start, 1); + q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + + msm_compr_set_volume(cstream, volume[0], volume[1]); + if (rc) + pr_err("%s : Set Volume failed : %d\n", + __func__, rc); + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("%s: SNDRV_PCM_TRIGGER_STOP\n", __func__); + atomic_set(&prtd->start, 0); + if (atomic_read(&prtd->eos)) { + prtd->cmd_interrupt = 1; + prtd->drain_ready = 1; + wake_up(&prtd->drain_wait); + wake_up(&prtd->eos_wait); + atomic_set(&prtd->eos, 0); + } + /* Issue flush command only if any buffers are left with DSP */ + spin_lock_irq(&prtd->lock); + if (prtd->bytes_received > prtd->copied_total) { + prtd->cmd_ack = 0; + spin_unlock_irq(&prtd->lock); + rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH); + if (rc < 0) { + pr_err("%s: flush cmd failed rc=%d\n", + __func__, rc); + return rc; + } + rc = wait_event_timeout(prtd->flush_wait, + prtd->cmd_ack, 1 * HZ); + if (!rc) + pr_err("Flush cmd timeout\n"); + } else + spin_unlock_irq(&prtd->lock); + + prtd->byte_offset = 0; + prtd->copied_total = 0; + prtd->app_pointer = 0; + prtd->bytes_received = 0; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH\n"); + q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + atomic_set(&prtd->start, 0); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("SNDRV_PCM_TRIGGER_PAUSE_RELEASE\n"); + atomic_set(&prtd->start, 1); + q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + break; + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + pr_debug("%s: SND_COMPR_TRIGGER_PARTIAL_DRAIN\n", __func__); + case SND_COMPR_TRIGGER_DRAIN: + pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__); + if (!atomic_read(&prtd->start)) { + pr_err("%s: stream is not in started state\n", + __func__); + break; + } + + /* Make sure all the data is sent to DSP before sending EOS */ + spin_lock_irq(&prtd->lock); + if (prtd->bytes_received > prtd->copied_total) { + atomic_set(&prtd->drain, 1); + prtd->drain_ready = 0; + spin_unlock_irq(&prtd->lock); + pr_debug("%s: wait till all the data is sent to dsp\n", + __func__); + rc = wait_event_interruptible(prtd->drain_wait, + prtd->drain_ready); + } else + spin_unlock_irq(&prtd->lock); + + if (!atomic_read(&prtd->start)) { + pr_err("%s: stream is not started\n", __func__); + break; + } + + if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN) + break; + + atomic_set(&prtd->eos, 1); + prtd->cmd_ack = 0; + pr_debug("%s: CMD_EOS\n", __func__); + q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + /* Wait indefinitely for DRAIN. Flush can also signal this*/ + rc = wait_event_interruptible(prtd->eos_wait, + (prtd->cmd_ack || prtd->cmd_interrupt)); + + if (rc < 0) + pr_err("%s: EOS cmd interrupted\n", __func__); + pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait\n", __func__); + + if (prtd->cmd_interrupt) + rc = -EINTR; + + prtd->cmd_interrupt = 0; + break; + case SND_COMPR_TRIGGER_NEXT_TRACK: + pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__); + break; + } + + return 0; +} + +static int msm_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *arg) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + struct snd_compr_tstamp tstamp; + uint64_t timestamp = 0; + int rc = 0; + + pr_debug("%s\n", __func__); + memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp)); + + spin_lock_irq(&prtd->lock); + tstamp.sampling_rate = prtd->sample_rate; + tstamp.byte_offset = prtd->byte_offset; + tstamp.copied_total = prtd->copied_total; + spin_unlock_irq(&prtd->lock); + + if (atomic_read(&prtd->start)) { + rc = q6asm_get_session_time(prtd->audio_client, ×tamp); + if (rc < 0) { + pr_err("%s: Get Session Time return value =%lld\n", + __func__, timestamp); + return -EAGAIN; + } + } + + /* DSP returns timestamp in usec */ + pr_debug("%s: timestamp = %lld usec\n", __func__, timestamp); + timestamp *= prtd->sample_rate; + tstamp.pcm_io_frames = (snd_pcm_uframes_t)div64_u64(timestamp, 1000000); + memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp)); + + return 0; +} + +static int msm_compr_ack(struct snd_compr_stream *cstream, + size_t count) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + void *src, *dstn; + size_t copy; + + pr_debug("%s: count = %d\n", __func__, count); + if (!prtd->buffer) { + pr_err("%s: Buffer is not allocated yet ??\n", __func__); + return -EINVAL; + } + src = runtime->buffer + prtd->app_pointer; + dstn = prtd->buffer + prtd->app_pointer; + if (count < prtd->buffer_size - prtd->app_pointer) { + memcpy(dstn, src, count); + prtd->app_pointer += count; + } else { + copy = prtd->buffer_size - prtd->app_pointer; + memcpy(dstn, src, copy); + memcpy(prtd->buffer, runtime->buffer, count - copy); + prtd->app_pointer = count - copy; + } + + /* + * If the stream is started and all the bytes received were + * copied to DSP, the newly received bytes should be + * sent right away + */ + spin_lock_irq(&prtd->lock); + + if (atomic_read(&prtd->start) && + prtd->bytes_received == prtd->copied_total) { + prtd->bytes_received += count; + msm_compr_send_buffer(prtd); + } else + prtd->bytes_received += count; + + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int msm_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + void *dstn; + size_t copy; + + pr_debug("%s: count = %d\n", __func__, count); + if (!prtd->buffer) { + pr_err("%s: Buffer is not allocated yet ??", __func__); + return 0; + } + + dstn = prtd->buffer + prtd->app_pointer; + if (count < prtd->buffer_size - prtd->app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + prtd->app_pointer += count; + } else { + copy = prtd->buffer_size - prtd->app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(prtd->buffer, buf + copy, count - copy)) + return -EFAULT; + prtd->app_pointer = count - copy; + } + + /* + * If stream is started and all the bytes received were + * copied to DSP, the newly received bytes should be + * copied right away + */ + spin_lock_irq(&prtd->lock); + + if (atomic_read(&prtd->start) && + prtd->bytes_received == prtd->copied_total) { + prtd->bytes_received += count; + msm_compr_send_buffer(prtd); + } else + prtd->bytes_received += count; + + spin_unlock_irq(&prtd->lock); + + return count; +} + +static int msm_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *arg) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + + pr_debug("%s\n", __func__); + memcpy(arg, &prtd->compr_cap, sizeof(struct snd_compr_caps)); + + return 0; +} + +static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + pr_debug("%s\n", __func__); + + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + codec->num_descriptors = 2; + codec->descriptor[0].max_ch = 2; + codec->descriptor[0].sample_rates = SNDRV_PCM_RATE_8000_48000; + codec->descriptor[0].bit_rate[0] = 320; /* 320kbps */ + codec->descriptor[0].bit_rate[1] = 128; + codec->descriptor[0].num_bitrates = 2; + codec->descriptor[0].profiles = 0; + codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO; + codec->descriptor[0].formats = 0; + break; + case SND_AUDIOCODEC_AAC: + codec->num_descriptors = 2; + codec->descriptor[1].max_ch = 2; + codec->descriptor[1].sample_rates = SNDRV_PCM_RATE_8000_48000; + codec->descriptor[1].bit_rate[0] = 320; /* 320kbps */ + codec->descriptor[1].bit_rate[1] = 128; + codec->descriptor[1].num_bitrates = 2; + codec->descriptor[1].profiles = 0; + codec->descriptor[1].modes = 0; + codec->descriptor[1].formats = + (SND_AUDIOSTREAMFORMAT_MP4ADTS | + SND_AUDIOSTREAMFORMAT_RAW); + break; + default: + pr_err("%s: Unsupported audio codec %d\n", + __func__, codec->codec); + return -EINVAL; + } + + return 0; +} + +static int msm_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + pr_debug("%s\n", __func__); + return -ENXIO; +} + +static int msm_compr_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) + snd_soc_platform_get_drvdata(platform); + struct snd_compr_stream *cstream = pdata->cstream[mc->reg]; + uint32_t *volume = pdata->volume[mc->reg]; + + volume[0] = ucontrol->value.integer.value[0]; + volume[1] = ucontrol->value.integer.value[1]; + pr_debug("%s: mc->reg %d left_vol %d right_vol %d\n", + __func__, mc->reg, volume[0], volume[1]); + if (cstream) + msm_compr_set_volume(cstream, volume[0], volume[1]); + return 0; +} + +static int msm_compr_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct msm_compr_pdata *pdata = + snd_soc_platform_get_drvdata(platform); + uint32_t *volume = pdata->volume[mc->reg]; + pr_debug("%s: mc->reg %d\n", __func__, mc->reg); + ucontrol->value.integer.value[0] = volume[0]; + ucontrol->value.integer.value[1] = volume[1]; + + return 0; +} + +/* System Pin has no volume control */ +static const struct snd_kcontrol_new msm_compr_volume_controls[] = { + SOC_DOUBLE_EXT_TLV("Compress Playback Volume", + MSM_FRONTEND_DAI_MULTIMEDIA4, + 0, 8, COMPRESSED_LR_VOL_MAX_STEPS, 0, + msm_compr_volume_get, + msm_compr_volume_put, + msm_compr_vol_gain), +}; + +static int msm_compr_probe(struct snd_soc_platform *platform) +{ + struct msm_compr_pdata *pdata; + int i; + + pr_debug("%s\n", __func__); + pdata = (struct msm_compr_pdata *) + kzalloc(sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + snd_soc_platform_set_drvdata(platform, pdata); + + atomic_set(&pdata->audio_ocmem_req, 0); + + for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) { + pdata->volume[i][0] = COMPRESSED_LR_VOL_MAX_STEPS; + pdata->volume[i][1] = COMPRESSED_LR_VOL_MAX_STEPS; + pdata->cstream[i] = NULL; + } + + return 0; +} + +static struct snd_compr_ops msm_compr_ops = { + .open = msm_compr_open, + .free = msm_compr_free, + .trigger = msm_compr_trigger, + .pointer = msm_compr_pointer, + .set_params = msm_compr_set_params, + .set_metadata = msm_compr_set_metadata, + .ack = msm_compr_ack, + .copy = msm_compr_copy, + .get_caps = msm_compr_get_caps, + .get_codec_caps = msm_compr_get_codec_caps, +}; + +static struct snd_soc_platform_driver msm_soc_platform = { + .probe = msm_compr_probe, + .compr_ops = &msm_compr_ops, + .controls = msm_compr_volume_controls, + .num_controls = ARRAY_SIZE(msm_compr_volume_controls), +}; + +static int msm_compr_dev_probe(struct platform_device *pdev) +{ + if (pdev->dev.of_node) + dev_set_name(&pdev->dev, "%s", "msm-compress-dsp"); + + pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev)); + return snd_soc_register_platform(&pdev->dev, + &msm_soc_platform); +} + +static int msm_compr_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static const struct of_device_id msm_compr_dt_match[] = { + {.compatible = "qcom,msm-compress-dsp"}, + {} +}; +MODULE_DEVICE_TABLE(of, msm_compr_dt_match); + +static struct platform_driver msm_compr_driver = { + .driver = { + .name = "msm-compress-dsp", + .owner = THIS_MODULE, + .of_match_table = msm_compr_dt_match, + }, + .probe = msm_compr_dev_probe, + .remove = msm_compr_remove, +}; + +static int __init msm_soc_platform_init(void) +{ + return platform_driver_register(&msm_compr_driver); +} +module_init(msm_soc_platform_init); + +static void __exit msm_soc_platform_exit(void) +{ + platform_driver_unregister(&msm_compr_driver); +} +module_exit(msm_soc_platform_exit); + +MODULE_DESCRIPTION("Compress Offload platform driver"); +MODULE_LICENSE("GPL v2");