Hi Luiz,
On 10 April 2014 10:25, Luiz Augusto von Dentz <luiz.dentz@xxxxxxxxx> wrote:
> Hi Andrzej,
>
> On Wed, Apr 9, 2014 at 5:16 PM, Andrzej Kaczmarek
> <andrzej.kaczmarek@xxxxxxxxx> wrote:
>> This patch adds new codec abstraction call which can be used to adjust
>> audio quality while playing. As for now we can either decrease quality
>> or restore default one.
>>
>> It's up to actual codec capabilities and implementation how this can be
>> handled. In case of SBC bitpool is decreased by fixed amount (5) until
>> min allowable value is reached (33) - the same values are used in
>> PulseAudio.
>> ---
>> android/hal-audio.c | 81 ++++++++++++++++++++++++++++++++++++++++++-----------
>> 1 file changed, 65 insertions(+), 16 deletions(-)
>>
>> diff --git a/android/hal-audio.c b/android/hal-audio.c
>> index e58be2b..2db927c 100644
>> --- a/android/hal-audio.c
>> +++ b/android/hal-audio.c
>> @@ -47,6 +47,9 @@
>>
>> #define MAX_DELAY 100000 /* 100ms */
>>
>> +#define SBC_QUALITY_MIN_BITPOOL 33
>> +#define SBC_QUALITY_STEP 5
>> +
>> static const uint8_t a2dp_src_uuid[] = {
>> 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
>> 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
>> @@ -128,6 +131,8 @@ struct sbc_data {
>>
>> sbc_t enc;
>>
>> + uint16_t payload_len;
>> +
>> size_t in_frame_len;
>> size_t in_buf_size;
>>
>> @@ -189,6 +194,10 @@ static size_t sbc_get_mediapacket_duration(void *codec_data);
>> static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
>> size_t len, struct media_packet *mp,
>> size_t mp_data_len, size_t *written);
>> +static bool sbc_quality_ctrl(void *codec_data, uint8_t op);
>> +
>> +#define QUALITY_CTRL_DEFAULT 0x00
>> +#define QUALITY_CTRL_DECREASE 0x01
>
> Lets call it QOS_POLICY_*
>
>> struct audio_codec {
>> uint8_t type;
>> @@ -205,6 +214,7 @@ struct audio_codec {
>> ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
>> size_t len, struct media_packet *mp,
>> size_t mp_data_len, size_t *written);
>> + bool (*quality_ctrl) (void *codec_data, uint8_t op);
>
> I prefer to use update_qos
>
>> };
>>
>> static const struct audio_codec audio_codecs[] = {
>> @@ -219,6 +229,7 @@ static const struct audio_codec audio_codecs[] = {
>> .get_buffer_size = sbc_get_buffer_size,
>> .get_mediapacket_duration = sbc_get_mediapacket_duration,
>> .encode_mediapacket = sbc_encode_mediapacket,
>> + .quality_ctrl = sbc_quality_ctrl,
>
> sbc_update_qos
>
>> }
>> };
>>
>> @@ -412,14 +423,33 @@ static void sbc_init_encoder(struct sbc_data *sbc_data)
>> in->min_bitpool, in->max_bitpool);
>> }
>>
>> -static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
>> - void **codec_data)
>> +static void sbc_codec_calculate(struct sbc_data *sbc_data)
>> {
>> - struct sbc_data *sbc_data;
>> size_t in_frame_len;
>> size_t out_frame_len;
>> size_t num_frames;
>>
>> + in_frame_len = sbc_get_codesize(&sbc_data->enc);
>> + out_frame_len = sbc_get_frame_length(&sbc_data->enc);
>> + num_frames = sbc_data->payload_len / out_frame_len;
>> +
>> + sbc_data->in_frame_len = in_frame_len;
>> + sbc_data->in_buf_size = num_frames * in_frame_len;
>> +
>> + sbc_data->out_frame_len = out_frame_len;
>> +
>> + sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
>> + sbc_data->frames_per_packet = num_frames;
>> +
>> + DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
>> + in_frame_len, out_frame_len, num_frames);
>> +}
>
> Looks like you remembered to update the frame size, however how about
> the latency? Does it gets updated automagically?
Latency is calculated using frame size on request, so as soon as
AudioFlinger calls out_get_latency it will get new value. I don't see
any callback to inform AF about the change but it does query for
latency from time to time so should be enough.
BR,
Andrzej
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