diff --git a/android/Android.mk b/android/Android.mk
index 4235a7c..f59352a 100644
--- a/android/Android.mk
+++ b/android/Android.mk
include $(CLEAR_VARS)
-LOCAL_SRC_FILES := bluez/android/hal-audio.c
+LOCAL_SRC_FILES := \
+ bluez/android/hal-audio.c \
+ bluez/android/hal-audio-sbc.c \
LOCAL_C_INCLUDES = \
$(call include-path-for, system-core) \
diff --git a/android/Makefile.am b/android/Makefile.am
index e663790..f2172a9 100644
--- a/android/Makefile.am
+++ b/android/Makefile.am
android_audio_a2dp_default_la_SOURCES = android/audio-msg.h \
android/hal-msg.h \
+ android/hal-audio.h \
android/hal-audio.c \
+ android/hal-audio-sbc.c \
android/hardware/audio.h \
android/hardware/audio_effect.h \
android/hardware/hardware.h \
diff --git a/android/hal-audio-sbc.c b/android/hal-audio-sbc.c
new file mode 100644
index 0000000..a16cf73
--- /dev/null
+++ b/android/hal-audio-sbc.c
+/*
+ * Copyright (C) 2013 Intel Corporation
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+#include <stdbool.h>
+#include <string.h>
+#include <malloc.h>
+
+#include <sbc/sbc.h>
+#include "audio-msg.h"
+#include "hal-audio.h"
+#include "hal-log.h"
+#include "../profiles/audio/a2dp-codecs.h"
+
+#define MAX_FRAMES_IN_PAYLOAD 15
+
+#define SBC_QUALITY_MIN_BITPOOL 33
+#define SBC_QUALITY_STEP 5
+
+struct sbc_data {
+ a2dp_sbc_t sbc;
+
+ sbc_t enc;
+
+ uint16_t payload_len;
+
+ size_t in_frame_len;
+ size_t in_buf_size;
+
+ size_t out_frame_len;
+
+ unsigned frame_duration;
+ unsigned frames_per_packet;
+};
+
+static const a2dp_sbc_t sbc_presets[] = {
+ {
+ .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
+ .channel_mode = SBC_CHANNEL_MODE_MONO |
+ SBC_CHANNEL_MODE_DUAL_CHANNEL |
+ SBC_CHANNEL_MODE_STEREO |
+ SBC_CHANNEL_MODE_JOINT_STEREO,
+ .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
+ .allocation_method = SBC_ALLOCATION_SNR |
+ SBC_ALLOCATION_LOUDNESS,
+ .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
+ SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
+ .min_bitpool = MIN_BITPOOL,
+ .max_bitpool = MAX_BITPOOL
+ },
+ {
+ .frequency = SBC_SAMPLING_FREQ_44100,
+ .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
+ .subbands = SBC_SUBBANDS_8,
+ .allocation_method = SBC_ALLOCATION_LOUDNESS,
+ .block_length = SBC_BLOCK_LENGTH_16,
+ .min_bitpool = MIN_BITPOOL,
+ .max_bitpool = MAX_BITPOOL
+ },
+ {
+ .frequency = SBC_SAMPLING_FREQ_48000,
+ .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
+ .subbands = SBC_SUBBANDS_8,
+ .allocation_method = SBC_ALLOCATION_LOUDNESS,
+ .block_length = SBC_BLOCK_LENGTH_16,
+ .min_bitpool = MIN_BITPOOL,
+ .max_bitpool = MAX_BITPOOL
+ },
+};
+
+static int sbc_get_presets(struct audio_preset *preset, size_t *len)
+{
+ int i;
+ int count;
+ size_t new_len = 0;
+ uint8_t *ptr = (uint8_t *) preset;
+ size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
+
+ count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
+
+ for (i = 0; i < count; i++) {
+ preset = (struct audio_preset *) ptr;
+
+ if (new_len + preset_size > *len)
+ break;
+
+ preset->len = sizeof(a2dp_sbc_t);
+ memcpy(preset->data, &sbc_presets[i], preset->len);
+
+ new_len += preset_size;
+ ptr += preset_size;
+ }
+
+ *len = new_len;
+
+ return i;
+}
+
+static int sbc_freq2int(uint8_t freq)
+{
+ switch (freq) {
+ case SBC_SAMPLING_FREQ_16000:
+ return 16000;
+ case SBC_SAMPLING_FREQ_32000:
+ return 32000;
+ case SBC_SAMPLING_FREQ_44100:
+ return 44100;
+ case SBC_SAMPLING_FREQ_48000:
+ return 48000;
+ default:
+ return 0;
+ }
+}
+
+static const char *sbc_mode2str(uint8_t mode)
+{
+ switch (mode) {
+ case SBC_CHANNEL_MODE_MONO:
+ return "Mono";
+ case SBC_CHANNEL_MODE_DUAL_CHANNEL:
+ return "DualChannel";
+ case SBC_CHANNEL_MODE_STEREO:
+ return "Stereo";
+ case SBC_CHANNEL_MODE_JOINT_STEREO:
+ return "JointStereo";
+ default:
+ return "(unknown)";
+ }
+}
+
+static int sbc_blocks2int(uint8_t blocks)
+{
+ switch (blocks) {
+ case SBC_BLOCK_LENGTH_4:
+ return 4;
+ case SBC_BLOCK_LENGTH_8:
+ return 8;
+ case SBC_BLOCK_LENGTH_12:
+ return 12;
+ case SBC_BLOCK_LENGTH_16:
+ return 16;
+ default:
+ return 0;
+ }
+}
+
+static int sbc_subbands2int(uint8_t subbands)
+{
+ switch (subbands) {
+ case SBC_SUBBANDS_4:
+ return 4;
+ case SBC_SUBBANDS_8:
+ return 8;
+ default:
+ return 0;
+ }
+}
+
+static const char *sbc_allocation2str(uint8_t allocation)
+{
+ switch (allocation) {
+ case SBC_ALLOCATION_SNR:
+ return "SNR";
+ case SBC_ALLOCATION_LOUDNESS:
+ return "Loudness";
+ default:
+ return "(unknown)";
+ }
+}
+
+static void sbc_init_encoder(struct sbc_data *sbc_data)
+{
+ a2dp_sbc_t *in = &sbc_data->sbc;
+ sbc_t *out = &sbc_data->enc;
+
+ sbc_init_a2dp(out, 0L, in, sizeof(*in));
+
+ out->endian = SBC_LE;
+ out->bitpool = in->max_bitpool;
+
+ DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d allocation=%s bitpool=%d-%d",
+ sbc_freq2int(in->frequency),
+ sbc_mode2str(in->channel_mode),
+ sbc_blocks2int(in->block_length),
+ sbc_subbands2int(in->subbands),
+ sbc_allocation2str(in->allocation_method),
+ in->min_bitpool, in->max_bitpool);
+}
+
+static void sbc_codec_calculate(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 bool 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 false;
+ }
+
+ sbc_data = calloc(sizeof(struct sbc_data), 1);
+ if (!sbc_data)
+ return false;
+
+ memcpy(&sbc_data->sbc, preset->data, preset->len);
+
+ sbc_init_encoder(sbc_data);
+
+ sbc_data->payload_len = payload_len;
+
+ sbc_codec_calculate(sbc_data);
+
+ *codec_data = sbc_data;
+
+ return true;
+}
+
+static bool sbc_cleanup(void *codec_data)
+{
+ struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
+
+ sbc_finish(&sbc_data->enc);
+ free(codec_data);
+
+ return true;
+}
+
+static bool sbc_get_config(void *codec_data, struct audio_input_config *config)
+{
+ struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
+
+ switch (sbc_data->sbc.frequency) {
+ case SBC_SAMPLING_FREQ_16000:
+ config->rate = 16000;
+ break;
+ case SBC_SAMPLING_FREQ_32000:
+ config->rate = 32000;
+ break;
+ case SBC_SAMPLING_FREQ_44100:
+ config->rate = 44100;
+ break;
+ case SBC_SAMPLING_FREQ_48000:
+ config->rate = 48000;
+ break;
+ default:
+ return false;
+ }
+ config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
+ AUDIO_CHANNEL_OUT_MONO :
+ AUDIO_CHANNEL_OUT_STEREO;
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+
+ return true;
+}
+
+static size_t sbc_get_buffer_size(void *codec_data)
+{
+ struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
+
+ return sbc_data->in_buf_size;
+}
+
+static size_t sbc_get_mediapacket_duration(void *codec_data)
+{
+ struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
+
+ return sbc_data->frame_duration * sbc_data->frames_per_packet;
+}
+
+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)
+{
+ struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
+ size_t consumed = 0;
+ size_t encoded = 0;
+ uint8_t frame_count = 0;
+
+ while (len - consumed >= sbc_data->in_frame_len &&
+ mp_data_len - encoded >= sbc_data->out_frame_len &&
+ frame_count < MAX_FRAMES_IN_PAYLOAD) {
+ ssize_t read;
+ ssize_t written = 0;
+
+ read = sbc_encode(&sbc_data->enc, buffer + consumed,
+ sbc_data->in_frame_len, mp->data + encoded,
+ mp_data_len - encoded, &written);
+
+ if (read < 0) {
+ error("SBC: failed to encode block at frame %d (%zd)",
+ frame_count, read);
+ break;
+ }
+
+ frame_count++;
+ consumed += read;
+ encoded += written;
+ }
+
+ *written = encoded;
+ mp->payload.frame_count = frame_count;
+
+ 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 const struct audio_codec codec = {
+ .type = A2DP_CODEC_SBC,
+
+ .get_presets = sbc_get_presets,
+
+ .init = sbc_codec_init,
+ .cleanup = sbc_cleanup,
+ .get_config = sbc_get_config,
+ .get_buffer_size = sbc_get_buffer_size,
+ .get_mediapacket_duration = sbc_get_mediapacket_duration,
+ .encode_mediapacket = sbc_encode_mediapacket,
+ .update_qos = sbc_update_qos,
+};
+
+const struct audio_codec *codec_sbc(void)
+{
+ return &codec;
+}
diff --git a/android/hal-audio.c b/android/hal-audio.c
index 1c889cc..df78497 100644
--- a/android/hal-audio.c
+++ b/android/hal-audio.c
#include <hardware/audio.h>
#include <hardware/hardware.h>
-#include <sbc/sbc.h>
-
#include "audio-msg.h"
#include "ipc-common.h"
#include "hal-log.h"
#include "hal-msg.h"
-#include "../profiles/audio/a2dp-codecs.h"
+#include "hal-audio.h"
#include "../src/shared/util.h"
#define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
#define FIXED_BUFFER_SIZE (20 * 512)
-#define MAX_FRAMES_IN_PAYLOAD 15
-
#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 };
static pthread_t ipc_th = 0;
static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
-#if __BYTE_ORDER == __LITTLE_ENDIAN
-
-struct rtp_header {
- unsigned cc:4;
- unsigned x:1;
- unsigned p:1;
- unsigned v:2;
-
- unsigned pt:7;
- unsigned m:1;
-
- uint16_t sequence_number;
- uint32_t timestamp;
- uint32_t ssrc;
- uint32_t csrc[0];
-} __attribute__ ((packed));
-
-struct rtp_payload {
- unsigned frame_count:4;
- unsigned rfa0:1;
- unsigned is_last_fragment:1;
- unsigned is_first_fragment:1;
- unsigned is_fragmented:1;
-} __attribute__ ((packed));
-
-#elif __BYTE_ORDER == __BIG_ENDIAN
-
-struct rtp_header {
- unsigned v:2;
- unsigned p:1;
- unsigned x:1;
- unsigned cc:4;
-
- unsigned m:1;
- unsigned pt:7;
-
- uint16_t sequence_number;
- uint32_t timestamp;
- uint32_t ssrc;
- uint32_t csrc[0];
-} __attribute__ ((packed));
-
-struct rtp_payload {
- unsigned is_fragmented:1;
- unsigned is_first_fragment:1;
- unsigned is_last_fragment:1;
- unsigned rfa0:1;
- unsigned frame_count:4;
-} __attribute__ ((packed));
-
-#else
-#error "Unknown byte order"
-#endif
-
-struct media_packet {
- struct rtp_header hdr;
- struct rtp_payload payload;
- uint8_t data[0];
-};
-
-struct audio_input_config {
- uint32_t rate;
- uint32_t channels;
- audio_format_t format;
-};
-
-struct sbc_data {
- a2dp_sbc_t sbc;
-
- sbc_t enc;
-
- uint16_t payload_len;
-
- size_t in_frame_len;
- size_t in_buf_size;
-
- size_t out_frame_len;
-
- unsigned frame_duration;
- unsigned frames_per_packet;
-};
-
static void timespec_add(struct timespec *base, uint64_t time_us,
struct timespec *res)
{
struct timespec *remain);
#endif
-static int sbc_get_presets(struct audio_preset *preset, size_t *len);
-static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
- void **codec_data);
-static int sbc_cleanup(void *codec_data);
-static int sbc_get_config(void *codec_data, struct audio_input_config *config);
-static size_t sbc_get_buffer_size(void *codec_data);
-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;
-
- int (*get_presets) (struct audio_preset *preset, size_t *len);
-
- int (*init) (struct audio_preset *preset, uint16_t mtu,
- void **codec_data);
- int (*cleanup) (void *codec_data);
- int (*get_config) (void *codec_data,
- struct audio_input_config *config);
- size_t (*get_buffer_size) (void *codec_data);
- size_t (*get_mediapacket_duration) (void *codec_data);
- 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[] = {
- {
- .type = A2DP_CODEC_SBC,
-
- .get_presets = sbc_get_presets,
-
- .init = sbc_codec_init,
- .cleanup = sbc_cleanup,
- .get_config = sbc_get_config,
- .get_buffer_size = sbc_get_buffer_size,
- .get_mediapacket_duration = sbc_get_mediapacket_duration,
- .encode_mediapacket = sbc_encode_mediapacket,
- .update_qos = sbc_update_qos,
- }
+static const audio_codec_get_t audio_codecs[] = {
+ codec_sbc,
};
#define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
struct a2dp_stream_out *out;
};
-static const a2dp_sbc_t sbc_presets[] = {
- {
- .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
- .channel_mode = SBC_CHANNEL_MODE_MONO |
- SBC_CHANNEL_MODE_DUAL_CHANNEL |
- SBC_CHANNEL_MODE_STEREO |
- SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_SNR |
- SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
- SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
- {
- .frequency = SBC_SAMPLING_FREQ_44100,
- .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
- {
- .frequency = SBC_SAMPLING_FREQ_48000,
- .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
-};
-
-static int sbc_get_presets(struct audio_preset *preset, size_t *len)
-{
- int i;
- int count;
- size_t new_len = 0;
- uint8_t *ptr = (uint8_t *) preset;
- size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
-
- count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
-
- for (i = 0; i < count; i++) {
- preset = (struct audio_preset *) ptr;
-
- if (new_len + preset_size > *len)
- break;
-
- preset->len = sizeof(a2dp_sbc_t);
- memcpy(preset->data, &sbc_presets[i], preset->len);
-
- new_len += preset_size;
- ptr += preset_size;
- }
-
- *len = new_len;
-
- return i;
-}
-
-static int sbc_freq2int(uint8_t freq)
-{
- switch (freq) {
- case SBC_SAMPLING_FREQ_16000:
- return 16000;
- case SBC_SAMPLING_FREQ_32000:
- return 32000;
- case SBC_SAMPLING_FREQ_44100:
- return 44100;
- case SBC_SAMPLING_FREQ_48000:
- return 48000;
- default:
- return 0;
- }
-}
-
-static const char *sbc_mode2str(uint8_t mode)
-{
- switch (mode) {
- case SBC_CHANNEL_MODE_MONO:
- return "Mono";
- case SBC_CHANNEL_MODE_DUAL_CHANNEL:
- return "DualChannel";
- case SBC_CHANNEL_MODE_STEREO:
- return "Stereo";
- case SBC_CHANNEL_MODE_JOINT_STEREO:
- return "JointStereo";
- default:
- return "(unknown)";
- }
-}
-
-static int sbc_blocks2int(uint8_t blocks)
-{
- switch (blocks) {
- case SBC_BLOCK_LENGTH_4:
- return 4;
- case SBC_BLOCK_LENGTH_8:
- return 8;
- case SBC_BLOCK_LENGTH_12:
- return 12;
- case SBC_BLOCK_LENGTH_16:
- return 16;
- default:
- return 0;
- }
-}
-
-static int sbc_subbands2int(uint8_t subbands)
-{
- switch (subbands) {
- case SBC_SUBBANDS_4:
- return 4;
- case SBC_SUBBANDS_8:
- return 8;
- default:
- return 0;
- }
-}
-
-static const char *sbc_allocation2str(uint8_t allocation)
-{
- switch (allocation) {
- case SBC_ALLOCATION_SNR:
- return "SNR";
- case SBC_ALLOCATION_LOUDNESS:
- return "Loudness";
- default:
- return "(unknown)";
- }
-}
-
-static void sbc_init_encoder(struct sbc_data *sbc_data)
-{
- a2dp_sbc_t *in = &sbc_data->sbc;
- sbc_t *out = &sbc_data->enc;
-
- sbc_init_a2dp(out, 0L, in, sizeof(*in));
-
- out->endian = SBC_LE;
- out->bitpool = in->max_bitpool;
-
- DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
- "allocation=%s bitpool=%d-%d",
- sbc_freq2int(in->frequency),
- sbc_mode2str(in->channel_mode),
- sbc_blocks2int(in->block_length),
- sbc_subbands2int(in->subbands),
- sbc_allocation2str(in->allocation_method),
- in->min_bitpool, in->max_bitpool);
-}
-
-static void sbc_codec_calculate(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;
- }
-
- sbc_data = calloc(sizeof(struct sbc_data), 1);
- if (!sbc_data)
- return AUDIO_STATUS_FAILED;
-
- memcpy(&sbc_data->sbc, preset->data, preset->len);
-
- sbc_init_encoder(sbc_data);
-
- sbc_data->payload_len = payload_len;
-
- sbc_codec_calculate(sbc_data);
-
- *codec_data = sbc_data;
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static int sbc_cleanup(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- sbc_finish(&sbc_data->enc);
- free(codec_data);
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static int sbc_get_config(void *codec_data, struct audio_input_config *config)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- switch (sbc_data->sbc.frequency) {
- case SBC_SAMPLING_FREQ_16000:
- config->rate = 16000;
- break;
- case SBC_SAMPLING_FREQ_32000:
- config->rate = 32000;
- break;
- case SBC_SAMPLING_FREQ_44100:
- config->rate = 44100;
- break;
- case SBC_SAMPLING_FREQ_48000:
- config->rate = 48000;
- break;
- default:
- return AUDIO_STATUS_FAILED;
- }
- config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
- AUDIO_CHANNEL_OUT_MONO :
- AUDIO_CHANNEL_OUT_STEREO;
- config->format = AUDIO_FORMAT_PCM_16_BIT;
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static size_t sbc_get_buffer_size(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- return sbc_data->in_buf_size;
-}
-
-static size_t sbc_get_mediapacket_duration(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- return sbc_data->frame_duration * sbc_data->frames_per_packet;
-}
-
-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)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
- size_t consumed = 0;
- size_t encoded = 0;
- uint8_t frame_count = 0;
-
- while (len - consumed >= sbc_data->in_frame_len &&
- mp_data_len - encoded >= sbc_data->out_frame_len &&
- frame_count < MAX_FRAMES_IN_PAYLOAD) {
- ssize_t read;
- ssize_t written = 0;
-
- read = sbc_encode(&sbc_data->enc, buffer + consumed,
- sbc_data->in_frame_len, mp->data + encoded,
- mp_data_len - encoded, &written);
-
- if (read < 0) {
- error("SBC: failed to encode block at frame %d (%zd)",
- frame_count, read);
- break;
- }
-
- frame_count++;
- consumed += read;
- encoded += written;
- }
-
- *written = encoded;
- mp->payload.frame_count = frame_count;
-
- 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)
{
size_t i;
for (i = 0; i < NUM_CODECS; i++, ep++) {
- const struct audio_codec *codec = &audio_codecs[i];
+ const struct audio_codec *codec = audio_codecs[i]();
+
+ if (!codec)
+ return AUDIO_STATUS_FAILED;
ep->id = ipc_open_cmd(codec);
diff --git a/android/hal-audio.h b/android/hal-audio.h
new file mode 100644
index 0000000..cc1a81c
--- /dev/null
+++ b/android/hal-audio.h
+/*
+ * Copyright (C) 2013 Intel Corporation
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+#include <time.h>
+#include <hardware/audio.h>
+
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+
+struct rtp_header {
+ unsigned cc:4;
+ unsigned x:1;
+ unsigned p:1;
+ unsigned v:2;
+
+ unsigned pt:7;
+ unsigned m:1;
+
+ uint16_t sequence_number;
+ uint32_t timestamp;
+ uint32_t ssrc;
+ uint32_t csrc[0];
+} __attribute__ ((packed));
+
+struct rtp_payload {
+ unsigned frame_count:4;
+ unsigned rfa0:1;
+ unsigned is_last_fragment:1;
+ unsigned is_first_fragment:1;
+ unsigned is_fragmented:1;
+} __attribute__ ((packed));
+
+#elif __BYTE_ORDER == __BIG_ENDIAN
+
+struct rtp_header {
+ unsigned v:2;
+ unsigned p:1;
+ unsigned x:1;
+ unsigned cc:4;
+
+ unsigned m:1;
+ unsigned pt:7;
+
+ uint16_t sequence_number;
+ uint32_t timestamp;
+ uint32_t ssrc;
+ uint32_t csrc[0];
+} __attribute__ ((packed));
+
+struct rtp_payload {
+ unsigned is_fragmented:1;
+ unsigned is_first_fragment:1;
+ unsigned is_last_fragment:1;
+ unsigned rfa0:1;
+ unsigned frame_count:4;
+} __attribute__ ((packed));
+
+#else
+#error "Unknown byte order"
+#endif
+
+struct media_packet {
+ struct rtp_header hdr;
+ struct rtp_payload payload;
+ uint8_t data[0];
+};
+
+struct audio_input_config {
+ uint32_t rate;
+ uint32_t channels;
+ audio_format_t format;
+};
+
+struct audio_codec {
+ uint8_t type;
+
+ int (*get_presets) (struct audio_preset *preset, size_t *len);
+
+ bool (*init) (struct audio_preset *preset, uint16_t mtu,
+ void **codec_data);
+ bool (*cleanup) (void *codec_data);
+ bool (*get_config) (void *codec_data,
+ struct audio_input_config *config);
+ size_t (*get_buffer_size) (void *codec_data);
+ size_t (*get_mediapacket_duration) (void *codec_data);
+ 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);
+};
+
+#define QOS_POLICY_DEFAULT 0x00
+#define QOS_POLICY_DECREASE 0x01
+
+typedef const struct audio_codec * (*audio_codec_get_t) (void);
+
+const struct audio_codec *codec_sbc(void);