From 80cb46c38507c8912026d0d53adf9782c97ba57a Mon Sep 17 00:00:00 2001
From: Eric Laurent <elaurent@google.com>
Date: Fri, 3 Jan 2014 18:30:14 -0800
Subject: [PATCH] ALSA: compress: change the way sample rates are sent to
kernel
The usage of SNDRV_RATES is not effective as we can have rates like 12000 or
some other ones used by decoders. This change the usage of this to use the raw
Hz values to be sent to kernel
Bug: 17398311.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Eric Laurent <elaurent@google.com>
---
include/uapi/sound/compress_params.h | 2 +-
sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c | 56 ++++++++--------------
2 files changed, 20 insertions(+), 38 deletions(-)
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 0fedc6a02b4c..25852a90bbbb 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -359,7 +359,7 @@ union snd_codec_options {
/** 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
+ * @sample_rates: Sampling rates in Hz, use values like 48000 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.
diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
index 57150b42e0a8..fbaa4bc68e4c 100644
--- a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
@@ -88,6 +88,10 @@ struct msm_compr_gapless_state {
bool use_dsp_gapless_mode;
};
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
struct msm_compr_pdata {
atomic_t audio_ocmem_req;
struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX];
@@ -909,46 +913,16 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
struct snd_compr_runtime *runtime = cstream->runtime;
struct msm_compr_audio *prtd = runtime->private_data;
int ret = 0, frame_sz = 0, delay_time_ms = 0;
+ int i, num_rates;
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;
- case SNDRV_PCM_RATE_96000:
- prtd->sample_rate = 96000;
- break;
- case SNDRV_PCM_RATE_192000:
- prtd->sample_rate = 192000;
- break;
- }
-
- pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate);
+ num_rates = sizeof(supported_sample_rates)/sizeof(unsigned int);
+ for (i = 0; i < num_rates; i++)
+ if (params->codec.sample_rate == supported_sample_rates[i])
+ break;
+ if (i == num_rates)
+ return -EINVAL;
prtd->compr_passthr = prtd->codec_param.codec.compr_passthr;
pr_debug("%s: compr_passthr = %d", __func__, prtd->compr_passthr);
@@ -1027,6 +1001,12 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0;
prtd->partial_drain_delay = delay_time_ms;
+ memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params));
+
+ /* ToDo: remove duplicates */
+ prtd->num_channels = prtd->codec_param.codec.ch_in;
+ prtd->sample_rate = prtd->codec_param.codec.sample_rate;
+ pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate);
ret = msm_compr_configure_dsp(cstream);
return ret;
@@ -1690,6 +1670,7 @@ static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream,
case SND_AUDIOCODEC_MP3:
codec->num_descriptors = 2;
codec->descriptor[0].max_ch = 2;
+ /* FIXME sample_rates in Hz */
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;
@@ -1701,6 +1682,7 @@ static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream,
case SND_AUDIOCODEC_AAC:
codec->num_descriptors = 2;
codec->descriptor[1].max_ch = 2;
+ /* FIXME sample_rates in Hz */
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;
--
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