diff options
Diffstat (limited to 'sound/soc/qcom/qdsp6')
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6adm.c | 7 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6afe.c | 8 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6afe.h | 1 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm-dai.c | 36 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm.c | 6 | ||||
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6routing.c | 2 |
6 files changed, 24 insertions, 36 deletions
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index da242515e146..2f3ea6beb066 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -403,7 +403,7 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, spin_lock_irqsave(&adm->copps_list_lock, flags); copp = q6adm_alloc_copp(adm, port_id); - if (IS_ERR_OR_NULL(copp)) { + if (IS_ERR(copp)) { spin_unlock_irqrestore(&adm->copps_list_lock, flags); return ERR_CAST(copp); } @@ -419,7 +419,6 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, copp->bit_width = bit_width; copp->app_type = app_type; - ret = q6adm_device_open(adm, copp, port_id, path, topology, channel_mode, bit_width, rate); if (ret < 0) { @@ -588,12 +587,12 @@ static int q6adm_probe(struct apr_device *adev) struct device *dev = &adev->dev; struct q6adm *adm; - adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); + adm = devm_kzalloc(dev, sizeof(*adm), GFP_KERNEL); if (!adm) return -ENOMEM; adm->apr = adev; - dev_set_drvdata(&adev->dev, adm); + dev_set_drvdata(dev, adm); adm->dev = dev; q6core_get_svc_api_info(adev->svc_id, &adm->ainfo); mutex_init(&adm->lock); diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 0ce4eb60f984..e0945f7a58c8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -800,14 +800,6 @@ int q6afe_get_port_id(int index) } EXPORT_SYMBOL_GPL(q6afe_get_port_id); -int q6afe_is_rx_port(int index) -{ - if (index < 0 || index >= AFE_PORT_MAX) - return -EINVAL; - - return port_maps[index].is_rx; -} -EXPORT_SYMBOL_GPL(q6afe_is_rx_port); static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, struct q6afe_port *port) { diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 1a0f80a14afe..c7ed5422baff 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -198,7 +198,6 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); -int q6afe_is_rx_port(int index); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index aff57052a735..9b7b218f2a20 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -37,9 +37,6 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 #define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) @@ -215,9 +212,10 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); struct q6asm_dai_rtd *prtd = runtime->private_data; struct q6asm_dai_data *pdata; + struct device *dev = component->dev; int ret, i; pdata = snd_soc_component_get_drvdata(component); @@ -225,7 +223,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, return -EINVAL; if (!prtd || !prtd->audio_client) { - pr_err("%s: private data null or audio client freed\n", + dev_err(dev, "%s: private data null or audio client freed\n", __func__); return -EINVAL; } @@ -248,7 +246,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, prtd->periods); if (ret < 0) { - pr_err("Audio Start: Buffer Allocation failed rc = %d\n", + dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n", ret); return -ENOMEM; } @@ -262,7 +260,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, } if (ret < 0) { - pr_err("%s: q6asm_open_write failed\n", __func__); + dev_err(dev, "%s: q6asm_open_write failed\n", __func__); q6asm_audio_client_free(prtd->audio_client); prtd->audio_client = NULL; return -ENOMEM; @@ -272,7 +270,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE, prtd->session_id, substream->stream); if (ret) { - pr_err("%s: stream reg failed ret:%d\n", __func__, ret); + dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret); return ret; } @@ -292,7 +290,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, } if (ret < 0) - pr_info("%s: CMD Format block failed\n", __func__); + dev_info(dev, "%s: CMD Format block failed\n", __func__); prtd->state = Q6ASM_STREAM_RUNNING; @@ -332,7 +330,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0); struct q6asm_dai_rtd *prtd; struct q6asm_dai_data *pdata; @@ -344,7 +342,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, pdata = snd_soc_component_get_drvdata(component); if (!pdata) { - pr_err("Drv data not found ..\n"); + dev_err(dev, "Drv data not found ..\n"); return -EINVAL; } @@ -357,7 +355,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); if (IS_ERR(prtd->audio_client)) { - pr_info("%s: Could not allocate memory\n", __func__); + dev_info(dev, "%s: Could not allocate memory\n", __func__); ret = PTR_ERR(prtd->audio_client); kfree(prtd); return ret; @@ -372,12 +370,12 @@ static int q6asm_dai_open(struct snd_soc_component *component, SNDRV_PCM_HW_PARAM_RATE, &constraints_sample_rates); if (ret < 0) - pr_info("snd_pcm_hw_constraint_list failed\n"); + dev_info(dev, "snd_pcm_hw_constraint_list failed\n"); /* Ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - pr_info("snd_pcm_hw_constraint_integer failed\n"); + dev_info(dev, "snd_pcm_hw_constraint_integer failed\n"); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = snd_pcm_hw_constraint_minmax(runtime, @@ -385,21 +383,21 @@ static int q6asm_dai_open(struct snd_soc_component *component, PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE, PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE); if (ret < 0) { - pr_err("constraint for buffer bytes min max ret = %d\n", - ret); + dev_err(dev, "constraint for buffer bytes min max ret = %d\n", + ret); } } ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret < 0) { - pr_err("constraint for period bytes step ret = %d\n", + dev_err(dev, "constraint for period bytes step ret = %d\n", ret); } ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret < 0) { - pr_err("constraint for buffer bytes step ret = %d\n", + dev_err(dev, "constraint for buffer bytes step ret = %d\n", ret); } @@ -424,7 +422,7 @@ static int q6asm_dai_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index ae4b2cabdf2d..755062eadcc8 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -311,7 +311,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac, 5 * HZ); if (!rc) { - dev_err(a->dev, "CMD timeout\n"); + dev_err(a->dev, "CMD %x timeout\n", hdr->opcode); rc = -ETIMEDOUT; } else if (ac->result.status > 0) { dev_err(a->dev, "DSP returned error[%x]\n", @@ -891,7 +891,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) rc = wait_event_timeout(ac->cmd_wait, (ac->result.opcode == hdr->opcode), 5 * HZ); if (!rc) { - dev_err(ac->dev, "CMD timeout\n"); + dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode); rc = -ETIMEDOUT; goto err; } @@ -912,9 +912,9 @@ err: /** * q6asm_open_write() - Open audio client for writing - * * @ac: audio client pointer * @format: audio sample format + * @codec_profile: compressed format profile * @bits_per_sample: bits per sample * * Return: Will be an negative value on error or zero on success diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 46e50612b92c..eaa95b5a7b66 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -924,7 +924,7 @@ static int routing_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct msm_routing_data *data = dev_get_drvdata(component->dev); unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id; struct session_data *session; |