android/hal-audio: Add support to control audio quality

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.

diff --git a/android/hal-audio.c b/android/hal-audio.c
index 3ff3b9d..8bfde7c 100644 (file)
--- 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_update_qos(void *codec_data, uint8_t op);
+
+#define QOS_POLICY_DEFAULT     0x00
+#define QOS_POLICY_DECREASE    0x01
 
 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 (*update_qos) (void *codec_data, uint8_t op);
 };
 
 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,
+               .update_qos = sbc_update_qos,
        }
 };
 
@@ -421,14 +432,33 @@ 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);
+}
+
+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;
@@ -442,20 +472,9 @@ 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;
 
@@ -550,6 +569,38 @@ static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
        return consumed;
 }
 
+static bool sbc_update_qos(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 QOS_POLICY_DEFAULT:
+               new_bitpool = sbc_data->sbc.max_bitpool;
+               break;
+
+       case QOS_POLICY_DECREASE:
+               if (curr_bitpool > SBC_QUALITY_MIN_BITPOOL) {
+                       new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
+                       if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
+                               new_bitpool = SBC_QUALITY_MIN_BITPOOL;
+               }
+               break;
+       }
+
+       if (new_bitpool == curr_bitpool)
+               return false;
+
+       sbc_data->enc.bitpool = new_bitpool;
+
+       sbc_codec_calculate(sbc_data);
+
+       info("SBC: bitpool changed: %d -> %d", curr_bitpool, new_bitpool);
+
+       return true;
+}
+
 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
                        void *param, size_t *rsp_len, void *rsp, int *fd)
 {