Hi Luiz,
On 11 April 2014 09:59, Luiz Augusto von Dentz <luiz.dentz@xxxxxxxxx> wrote:
> Hi Andrzej,
>
> On Thu, Apr 10, 2014 at 11:25 AM, 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?
>>
>>> +static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
>>> + void **codec_data)
>>> +{
>>> + struct sbc_data *sbc_data;
>>> +
>>> if (preset->len != sizeof(a2dp_sbc_t)) {
>>> error("SBC: preset size mismatch");
>>> return AUDIO_STATUS_FAILED;
>>> @@ -433,20 +463,9 @@ static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
>>>
>>> sbc_init_encoder(sbc_data);
>>>
>>> - in_frame_len = sbc_get_codesize(&sbc_data->enc);
>>> - out_frame_len = sbc_get_frame_length(&sbc_data->enc);
>>> - num_frames = 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;
>>> + sbc_data->payload_len = payload_len;
>>>
>>> - DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
>>> - in_frame_len, out_frame_len, num_frames);
>>> + sbc_codec_calculate(sbc_data);
>>>
>>> *codec_data = sbc_data;
>>>
>>> @@ -541,6 +560,36 @@ static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
>>> return consumed;
>>> }
>>>
>>> +static bool sbc_quality_ctrl(void *codec_data, uint8_t op)
>>> +{
>>> + struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
>>> + uint8_t curr_bitpool = sbc_data->enc.bitpool;
>>> + uint8_t new_bitpool = curr_bitpool;
>>> +
>>> + switch (op) {
>>> + case QUALITY_CTRL_DEFAULT:
>>> + new_bitpool = sbc_data->sbc.max_bitpool;
>>> + break;
>>> +
>>> + case QUALITY_CTRL_DECREASE:
>>> + new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
>>> + if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
>>> + new_bitpool = SBC_QUALITY_MIN_BITPOOL;
>>> + break;
>>> + }
>
> One detail is missing here, if max_bitpool is bellow the
> SBC_QUALITY_MIN_BITPOOL you should do nothing otherwise we might send
> frames using a bitpool outside of the negotiated range.
I changed this to try to decrease bitpool only if we're above
SBC_QUALITY_MIN_BITPOOL so if we are already below, we won't touch
bitpool.
BR,
Andrzej
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