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-rw-r--r--audio/audio_aec.c700
1 files changed, 700 insertions, 0 deletions
diff --git a/audio/audio_aec.c b/audio/audio_aec.c
new file mode 100644
index 0000000..ab99c93
--- /dev/null
+++ b/audio/audio_aec.c
@@ -0,0 +1,700 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * 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.
+ */
+
+// clang-format off
+/*
+ * Typical AEC signal flow:
+ *
+ * Microphone Audio
+ * Timestamps
+ * +--------------------------------------+
+ * | | +---------------+
+ * | Microphone +---------------+ | | |
+ * O|====== | Audio | Sample Rate | +-------> |
+ * (from . +--+ Samples | + | | |
+ * mic . +==================> Format |==============> |
+ * codec) . | Conversion | | | Cleaned
+ * O|====== | (if required) | | Acoustic | Audio
+ * +---------------+ | Echo | Samples
+ * | Canceller |===================>
+ * | (AEC) |
+ * Reference +---------------+ | |
+ * Audio | Sample Rate | | |
+ * Samples | + | | |
+ * +=============> Format |==============> |
+ * | | Conversion | | |
+ * | | (if required) | +-------> |
+ * | +---------------+ | | |
+ * | | +---------------+
+ * | +-------------------------------+
+ * | | Reference Audio
+ * | | Timestamps
+ * | |
+ * +--+----+---------+ AUDIO CAPTURE
+ * | Speaker |
+ * +------------+ Audio/Timestamp +---------------------------------------------------------------------------+
+ * | Buffer |
+ * +--^----^---------+ AUDIO PLAYBACK
+ * | |
+ * | |
+ * | |
+ * | |
+ * |\ | |
+ * | +-+ | |
+ * (to | | +-----C----+
+ * speaker | | | | Playback
+ * codec) | | <=====+================================================================+ Audio
+ * | +-+ Samples
+ * |/
+ *
+ */
+// clang-format on
+
+#define LOG_TAG "audio_hw_aec"
+// #define LOG_NDEBUG 0
+
+#include <audio_utils/primitives.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <errno.h>
+#include <malloc.h>
+#include <sys/time.h>
+#include <tinyalsa/asoundlib.h>
+#include <unistd.h>
+#include <log/log.h>
+#include "audio_aec.h"
+
+#ifdef AEC_HAL
+#include "audio_aec_process.h"
+#else
+#define aec_spk_mic_init(...) ((int)0)
+#define aec_spk_mic_reset(...) ((void)0)
+#define aec_spk_mic_process(...) ((int32_t)0)
+#define aec_spk_mic_release(...) ((void)0)
+#endif
+
+#define MAX_TIMESTAMP_DIFF_USEC 200000
+
+#define MAX_READ_WAIT_TIME_MSEC 80
+
+uint64_t timespec_to_usec(struct timespec ts) {
+ return (ts.tv_sec * 1e6L + ts.tv_nsec/1000);
+}
+
+void get_reference_audio_in_place(struct aec_t *aec, size_t frames) {
+ if (aec->num_reference_channels == aec->spk_num_channels) {
+ /* Reference count equals speaker channels, nothing to do here. */
+ return;
+ } else if (aec->num_reference_channels != 1) {
+ /* We don't have a rule for non-mono references, show error on log */
+ ALOGE("Invalid reference count - must be 1 or match number of playback channels!");
+ return;
+ }
+ int16_t *src_Nch = &aec->spk_buf_playback_format[0];
+ int16_t *dst_1ch = &aec->spk_buf_playback_format[0];
+ int32_t num_channels = (int32_t)aec->spk_num_channels;
+ size_t frame, ch;
+ for (frame = 0; frame < frames; frame++) {
+ int32_t acc = 0;
+ for (ch = 0; ch < aec->spk_num_channels; ch++) {
+ acc += src_Nch[ch];
+ }
+ *dst_1ch++ = clamp16(acc/num_channels);
+ src_Nch += aec->spk_num_channels;
+ }
+}
+
+void print_queue_status_to_log(struct aec_t *aec, bool write_side) {
+ ssize_t q1 = fifo_available_to_read(aec->spk_fifo);
+ ssize_t q2 = fifo_available_to_read(aec->ts_fifo);
+
+ ALOGV("Queue available %s: Spk %zd (count %zd) TS %zd (count %zd)",
+ (write_side) ? "(POST-WRITE)" : "(PRE-READ)",
+ q1, q1/aec->spk_frame_size_bytes/PLAYBACK_PERIOD_SIZE,
+ q2, q2/sizeof(struct aec_info));
+}
+
+void flush_aec_fifos(struct aec_t *aec) {
+ if (aec == NULL) {
+ return;
+ }
+ if (aec->spk_fifo != NULL) {
+ ALOGV("Flushing AEC Spk FIFO...");
+ fifo_flush(aec->spk_fifo);
+ }
+ if (aec->ts_fifo != NULL) {
+ ALOGV("Flushing AEC Timestamp FIFO...");
+ fifo_flush(aec->ts_fifo);
+ }
+ /* Reset FIFO read-write offset tracker */
+ aec->read_write_diff_bytes = 0;
+}
+
+void aec_set_spk_running_no_lock(struct aec_t* aec, bool state) {
+ aec->spk_running = state;
+}
+
+bool aec_get_spk_running_no_lock(struct aec_t* aec) {
+ return aec->spk_running;
+}
+
+void destroy_aec_reference_config_no_lock(struct aec_t* aec) {
+ if (!aec->spk_initialized) {
+ return;
+ }
+ aec_set_spk_running_no_lock(aec, false);
+ fifo_release(aec->spk_fifo);
+ fifo_release(aec->ts_fifo);
+ memset(&aec->last_spk_info, 0, sizeof(struct aec_info));
+ aec->spk_initialized = false;
+}
+
+void destroy_aec_mic_config_no_lock(struct aec_t* aec) {
+ if (!aec->mic_initialized) {
+ return;
+ }
+ release_resampler(aec->spk_resampler);
+ free(aec->mic_buf);
+ free(aec->spk_buf);
+ free(aec->spk_buf_playback_format);
+ free(aec->spk_buf_resampler_out);
+ memset(&aec->last_mic_info, 0, sizeof(struct aec_info));
+ aec->mic_initialized = false;
+}
+
+struct aec_t *init_aec_interface() {
+ ALOGV("%s enter", __func__);
+ struct aec_t *aec = (struct aec_t *)calloc(1, sizeof(struct aec_t));
+ if (aec == NULL) {
+ ALOGE("Failed to allocate memory for AEC interface!");
+ } else {
+ pthread_mutex_init(&aec->lock, NULL);
+ }
+
+ ALOGV("%s exit", __func__);
+ return aec;
+}
+
+void release_aec_interface(struct aec_t *aec) {
+ ALOGV("%s enter", __func__);
+ pthread_mutex_lock(&aec->lock);
+ destroy_aec_mic_config_no_lock(aec);
+ destroy_aec_reference_config_no_lock(aec);
+ pthread_mutex_unlock(&aec->lock);
+ free(aec);
+ ALOGV("%s exit", __func__);
+}
+
+int init_aec(int sampling_rate, int num_reference_channels,
+ int num_microphone_channels, struct aec_t **aec_ptr) {
+ ALOGV("%s enter", __func__);
+ int ret = 0;
+ int aec_ret = aec_spk_mic_init(
+ sampling_rate,
+ num_reference_channels,
+ num_microphone_channels);
+ if (aec_ret) {
+ ALOGE("AEC object failed to initialize!");
+ ret = -EINVAL;
+ }
+ struct aec_t *aec = init_aec_interface();
+ if (!ret) {
+ aec->num_reference_channels = num_reference_channels;
+ /* Set defaults, will be overridden by settings in init_aec_(mic|referece_config) */
+ /* Capture uses 2-ch, 32-bit frames */
+ aec->mic_sampling_rate = CAPTURE_CODEC_SAMPLING_RATE;
+ aec->mic_frame_size_bytes = CHANNEL_STEREO * sizeof(int32_t);
+ aec->mic_num_channels = CHANNEL_STEREO;
+
+ /* Playback uses 2-ch, 16-bit frames */
+ aec->spk_sampling_rate = PLAYBACK_CODEC_SAMPLING_RATE;
+ aec->spk_frame_size_bytes = CHANNEL_STEREO * sizeof(int16_t);
+ aec->spk_num_channels = CHANNEL_STEREO;
+ }
+
+ (*aec_ptr) = aec;
+ ALOGV("%s exit", __func__);
+ return ret;
+}
+
+void release_aec(struct aec_t *aec) {
+ ALOGV("%s enter", __func__);
+ if (aec == NULL) {
+ return;
+ }
+ release_aec_interface(aec);
+ aec_spk_mic_release();
+ ALOGV("%s exit", __func__);
+}
+
+int init_aec_reference_config(struct aec_t *aec, struct alsa_stream_out *out) {
+ ALOGV("%s enter", __func__);
+ if (!aec) {
+ ALOGE("AEC: No valid interface found!");
+ return -EINVAL;
+ }
+
+ int ret = 0;
+ pthread_mutex_lock(&aec->lock);
+ if (aec->spk_initialized) {
+ destroy_aec_reference_config_no_lock(aec);
+ }
+
+ aec->spk_fifo = fifo_init(
+ out->config.period_count * out->config.period_size *
+ audio_stream_out_frame_size(&out->stream),
+ false /* reader_throttles_writer */);
+ if (aec->spk_fifo == NULL) {
+ ALOGE("AEC: Speaker loopback FIFO Init failed!");
+ ret = -EINVAL;
+ goto exit;
+ }
+ aec->ts_fifo = fifo_init(
+ out->config.period_count * sizeof(struct aec_info),
+ false /* reader_throttles_writer */);
+ if (aec->ts_fifo == NULL) {
+ ALOGE("AEC: Speaker timestamp FIFO Init failed!");
+ ret = -EINVAL;
+ fifo_release(aec->spk_fifo);
+ goto exit;
+ }
+
+ aec->spk_sampling_rate = out->config.rate;
+ aec->spk_frame_size_bytes = audio_stream_out_frame_size(&out->stream);
+ aec->spk_num_channels = out->config.channels;
+ aec->spk_initialized = true;
+exit:
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+ return ret;
+}
+
+void destroy_aec_reference_config(struct aec_t* aec) {
+ ALOGV("%s enter", __func__);
+ if (aec == NULL) {
+ ALOGV("%s exit", __func__);
+ return;
+ }
+ pthread_mutex_lock(&aec->lock);
+ destroy_aec_reference_config_no_lock(aec);
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+}
+
+int write_to_reference_fifo(struct aec_t* aec, void* buffer, struct aec_info* info) {
+ ALOGV("%s enter", __func__);
+ int ret = 0;
+ size_t bytes = info->bytes;
+
+ /* Write audio samples to FIFO */
+ ssize_t written_bytes = fifo_write(aec->spk_fifo, buffer, bytes);
+ if (written_bytes != bytes) {
+ ALOGE("Could only write %zu of %zu bytes", written_bytes, bytes);
+ ret = -ENOMEM;
+ }
+
+ /* Write timestamp to FIFO */
+ info->bytes = written_bytes;
+ ALOGV("Speaker timestamp: %ld s, %ld nsec", info->timestamp.tv_sec, info->timestamp.tv_nsec);
+ ssize_t ts_bytes = fifo_write(aec->ts_fifo, info, sizeof(struct aec_info));
+ ALOGV("Wrote TS bytes: %zu", ts_bytes);
+ print_queue_status_to_log(aec, true);
+ ALOGV("%s exit", __func__);
+ return ret;
+}
+
+void get_spk_timestamp(struct aec_t* aec, ssize_t read_bytes, uint64_t* spk_time) {
+ *spk_time = 0;
+ uint64_t spk_time_offset = 0;
+ float usec_per_byte = 1E6 / ((float)(aec->spk_frame_size_bytes * aec->spk_sampling_rate));
+ if (aec->read_write_diff_bytes < 0) {
+ /* We're still reading a previous write packet. (We only need the first sample's timestamp,
+ * so even if we straddle packets we only care about the first one)
+ * So we just use the previous timestamp, with an appropriate offset
+ * based on the number of bytes remaining to be read from that write packet. */
+ spk_time_offset = (aec->last_spk_info.bytes + aec->read_write_diff_bytes) * usec_per_byte;
+ ALOGV("Reusing previous timestamp, calculated offset (usec) %" PRIu64, spk_time_offset);
+ } else {
+ /* If read_write_diff_bytes > 0, there are no new writes, so there won't be timestamps in
+ * the FIFO, and the check below will fail. */
+ if (!fifo_available_to_read(aec->ts_fifo)) {
+ ALOGE("Timestamp error: no new timestamps!");
+ return;
+ }
+ /* We just read valid data, so if we're here, we should have a valid timestamp to use. */
+ ssize_t ts_bytes = fifo_read(aec->ts_fifo, &aec->last_spk_info, sizeof(struct aec_info));
+ ALOGV("Read TS bytes: %zd, expected %zu", ts_bytes, sizeof(struct aec_info));
+ aec->read_write_diff_bytes -= aec->last_spk_info.bytes;
+ }
+
+ *spk_time = timespec_to_usec(aec->last_spk_info.timestamp) + spk_time_offset;
+
+ aec->read_write_diff_bytes += read_bytes;
+ struct aec_info spk_info = aec->last_spk_info;
+ while (aec->read_write_diff_bytes > 0) {
+ /* If read_write_diff_bytes > 0, it means that there are more write packet timestamps
+ * in FIFO (since there we read more valid data the size of the current timestamp's
+ * packet). Keep reading timestamps from FIFO to get to the most recent one. */
+ if (!fifo_available_to_read(aec->ts_fifo)) {
+ /* There are no more timestamps, we have the most recent one. */
+ ALOGV("At the end of timestamp FIFO, breaking...");
+ break;
+ }
+ fifo_read(aec->ts_fifo, &spk_info, sizeof(struct aec_info));
+ ALOGV("Fast-forwarded timestamp by %zd bytes, remaining bytes: %zd,"
+ " new timestamp (usec) %" PRIu64,
+ spk_info.bytes, aec->read_write_diff_bytes, timespec_to_usec(spk_info.timestamp));
+ aec->read_write_diff_bytes -= spk_info.bytes;
+ }
+ aec->last_spk_info = spk_info;
+}
+
+int get_reference_samples(struct aec_t* aec, void* buffer, struct aec_info* info) {
+ ALOGV("%s enter", __func__);
+
+ if (!aec->spk_initialized) {
+ ALOGE("%s called with no reference initialized", __func__);
+ return -EINVAL;
+ }
+
+ size_t bytes = info->bytes;
+ const size_t frames = bytes / aec->mic_frame_size_bytes;
+ const size_t sample_rate_ratio = aec->spk_sampling_rate / aec->mic_sampling_rate;
+
+ /* Read audio samples from FIFO */
+ const size_t req_bytes = frames * sample_rate_ratio * aec->spk_frame_size_bytes;
+ ssize_t available_bytes = 0;
+ unsigned int wait_count = MAX_READ_WAIT_TIME_MSEC;
+ while (true) {
+ available_bytes = fifo_available_to_read(aec->spk_fifo);
+ if (available_bytes >= req_bytes) {
+ break;
+ } else if (available_bytes < 0) {
+ ALOGE("fifo_read returned code %zu ", available_bytes);
+ return -ENOMEM;
+ }
+
+ ALOGV("Sleeping, required bytes: %zu, available bytes: %zd", req_bytes, available_bytes);
+ usleep(1000);
+ if ((wait_count--) == 0) {
+ ALOGE("Timed out waiting for read from reference FIFO");
+ return -ETIMEDOUT;
+ }
+ }
+
+ const size_t read_bytes = fifo_read(aec->spk_fifo, aec->spk_buf_playback_format, req_bytes);
+
+ /* Get timestamp*/
+ get_spk_timestamp(aec, read_bytes, &info->timestamp_usec);
+
+ /* Get reference - could be mono, downmixed from multichannel.
+ * Reference stored at spk_buf_playback_format */
+ const size_t resampler_in_frames = frames * sample_rate_ratio;
+ get_reference_audio_in_place(aec, resampler_in_frames);
+
+ int16_t* resampler_out_buf;
+ /* Resample to mic sampling rate (16-bit resampler) */
+ if (aec->spk_resampler != NULL) {
+ size_t in_frame_count = resampler_in_frames;
+ size_t out_frame_count = frames;
+ aec->spk_resampler->resample_from_input(aec->spk_resampler, aec->spk_buf_playback_format,
+ &in_frame_count, aec->spk_buf_resampler_out,
+ &out_frame_count);
+ resampler_out_buf = aec->spk_buf_resampler_out;
+ } else {
+ if (sample_rate_ratio != 1) {
+ ALOGE("Speaker sample rate %d, mic sample rate %d but no resampler defined!",
+ aec->spk_sampling_rate, aec->mic_sampling_rate);
+ }
+ resampler_out_buf = aec->spk_buf_playback_format;
+ }
+
+ /* Convert to 32 bit */
+ int16_t* src16 = resampler_out_buf;
+ int32_t* dst32 = buffer;
+ size_t frame, ch;
+ for (frame = 0; frame < frames; frame++) {
+ for (ch = 0; ch < aec->num_reference_channels; ch++) {
+ *dst32++ = ((int32_t)*src16++) << 16;
+ }
+ }
+
+ info->bytes = bytes;
+
+ ALOGV("%s exit", __func__);
+ return 0;
+}
+
+int init_aec_mic_config(struct aec_t *aec, struct alsa_stream_in *in) {
+ ALOGV("%s enter", __func__);
+#if DEBUG_AEC
+ remove("/data/local/traces/aec_in.pcm");
+ remove("/data/local/traces/aec_out.pcm");
+ remove("/data/local/traces/aec_ref.pcm");
+ remove("/data/local/traces/aec_timestamps.txt");
+#endif /* #if DEBUG_AEC */
+
+ if (!aec) {
+ ALOGE("AEC: No valid interface found!");
+ return -EINVAL;
+ }
+
+ int ret = 0;
+ pthread_mutex_lock(&aec->lock);
+ if (aec->mic_initialized) {
+ destroy_aec_mic_config_no_lock(aec);
+ }
+ aec->mic_sampling_rate = in->config.rate;
+ aec->mic_frame_size_bytes = audio_stream_in_frame_size(&in->stream);
+ aec->mic_num_channels = in->config.channels;
+
+ aec->mic_buf_size_bytes = in->config.period_size * audio_stream_in_frame_size(&in->stream);
+ aec->mic_buf = (int32_t *)malloc(aec->mic_buf_size_bytes);
+ if (aec->mic_buf == NULL) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ memset(aec->mic_buf, 0, aec->mic_buf_size_bytes);
+ /* Reference buffer is the same number of frames as mic,
+ * only with a different number of channels in the frame. */
+ aec->spk_buf_size_bytes = in->config.period_size * aec->spk_frame_size_bytes;
+ aec->spk_buf = (int32_t *)malloc(aec->spk_buf_size_bytes);
+ if (aec->spk_buf == NULL) {
+ ret = -ENOMEM;
+ goto exit_1;
+ }
+ memset(aec->spk_buf, 0, aec->spk_buf_size_bytes);
+
+ /* Pre-resampler buffer */
+ size_t spk_frame_out_format_bytes = aec->spk_sampling_rate / aec->mic_sampling_rate *
+ aec->spk_buf_size_bytes;
+ aec->spk_buf_playback_format = (int16_t *)malloc(spk_frame_out_format_bytes);
+ if (aec->spk_buf_playback_format == NULL) {
+ ret = -ENOMEM;
+ goto exit_2;
+ }
+ /* Resampler is 16-bit */
+ aec->spk_buf_resampler_out = (int16_t *)malloc(aec->spk_buf_size_bytes);
+ if (aec->spk_buf_resampler_out == NULL) {
+ ret = -ENOMEM;
+ goto exit_3;
+ }
+
+ /* Don't use resampler if it's not required */
+ if (in->config.rate == aec->spk_sampling_rate) {
+ aec->spk_resampler = NULL;
+ } else {
+ int resampler_ret = create_resampler(
+ aec->spk_sampling_rate, in->config.rate, aec->num_reference_channels,
+ RESAMPLER_QUALITY_MAX - 1, /* MAX - 1 is the real max */
+ NULL, /* resampler_buffer_provider */
+ &aec->spk_resampler);
+ if (resampler_ret) {
+ ALOGE("AEC: Resampler initialization failed! Error code %d", resampler_ret);
+ ret = resampler_ret;
+ goto exit_4;
+ }
+ }
+
+ flush_aec_fifos(aec);
+ aec_spk_mic_reset();
+ aec->mic_initialized = true;
+
+exit:
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+ return ret;
+
+exit_4:
+ free(aec->spk_buf_resampler_out);
+exit_3:
+ free(aec->spk_buf_playback_format);
+exit_2:
+ free(aec->spk_buf);
+exit_1:
+ free(aec->mic_buf);
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+ return ret;
+}
+
+void aec_set_spk_running(struct aec_t *aec, bool state) {
+ ALOGV("%s enter", __func__);
+ pthread_mutex_lock(&aec->lock);
+ aec_set_spk_running_no_lock(aec, state);
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+}
+
+bool aec_get_spk_running(struct aec_t *aec) {
+ ALOGV("%s enter", __func__);
+ pthread_mutex_lock(&aec->lock);
+ bool state = aec_get_spk_running_no_lock(aec);
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+ return state;
+}
+
+void destroy_aec_mic_config(struct aec_t* aec) {
+ ALOGV("%s enter", __func__);
+ if (aec == NULL) {
+ ALOGV("%s exit", __func__);
+ return;
+ }
+
+ pthread_mutex_lock(&aec->lock);
+ destroy_aec_mic_config_no_lock(aec);
+ pthread_mutex_unlock(&aec->lock);
+ ALOGV("%s exit", __func__);
+}
+
+#ifdef AEC_HAL
+int process_aec(struct aec_t *aec, void* buffer, struct aec_info *info) {
+ ALOGV("%s enter", __func__);
+ int ret = 0;
+
+ if (aec == NULL) {
+ ALOGE("AEC: Interface uninitialized! Cannot process.");
+ return -EINVAL;
+ }
+
+ if ((!aec->mic_initialized) || (!aec->spk_initialized)) {
+ ALOGE("%s called with initialization: mic: %d, spk: %d", __func__, aec->mic_initialized,
+ aec->spk_initialized);
+ return -EINVAL;
+ }
+
+ size_t bytes = info->bytes;
+
+ size_t frame_size = aec->mic_frame_size_bytes;
+ size_t in_frames = bytes / frame_size;
+
+ /* Copy raw mic samples to AEC input buffer */
+ memcpy(aec->mic_buf, buffer, bytes);
+
+ uint64_t mic_time = timespec_to_usec(info->timestamp);
+ uint64_t spk_time = 0;
+
+ /*
+ * Only run AEC if there is speaker playback.
+ * The first time speaker state changes to running, flush FIFOs, so we're not stuck
+ * processing stale reference input.
+ */
+ bool spk_running = aec_get_spk_running(aec);
+
+ if (!spk_running) {
+ /* No new playback samples, so don't run AEC.
+ * 'buffer' already contains input samples. */
+ ALOGV("Speaker not running, skipping AEC..");
+ goto exit;
+ }
+
+ if (!aec->prev_spk_running) {
+ flush_aec_fifos(aec);
+ }
+
+ /* If there's no data in FIFO, exit */
+ if (fifo_available_to_read(aec->spk_fifo) <= 0) {
+ ALOGV("Echo reference buffer empty, zeroing reference....");
+ goto exit;
+ }
+
+ print_queue_status_to_log(aec, false);
+
+ /* Get reference, with format and sample rate required by AEC */
+ struct aec_info spk_info;
+ spk_info.bytes = bytes;
+ int ref_ret = get_reference_samples(aec, aec->spk_buf, &spk_info);
+ spk_time = spk_info.timestamp_usec;
+
+ if (ref_ret) {
+ ALOGE("get_reference_samples returned code %d", ref_ret);
+ ret = -ENOMEM;
+ goto exit;
+ }
+
+ int64_t time_diff = (mic_time > spk_time) ? (mic_time - spk_time) : (spk_time - mic_time);
+ if ((spk_time == 0) || (mic_time == 0) || (time_diff > MAX_TIMESTAMP_DIFF_USEC)) {
+ ALOGV("Speaker-mic timestamps diverged, skipping AEC");
+ flush_aec_fifos(aec);
+ aec_spk_mic_reset();
+ goto exit;
+ }
+
+ ALOGV("Mic time: %"PRIu64", spk time: %"PRIu64, mic_time, spk_time);
+
+ /*
+ * AEC processing call - output stored at 'buffer'
+ */
+ int32_t aec_status = aec_spk_mic_process(
+ aec->spk_buf, spk_time,
+ aec->mic_buf, mic_time,
+ in_frames,
+ buffer);
+
+ if (!aec_status) {
+ ALOGE("AEC processing failed!");
+ ret = -EINVAL;
+ }
+
+exit:
+ aec->prev_spk_running = spk_running;
+ ALOGV("Mic time: %"PRIu64", spk time: %"PRIu64, mic_time, spk_time);
+ if (ret) {
+ /* Best we can do is copy over the raw mic signal */
+ memcpy(buffer, aec->mic_buf, bytes);
+ flush_aec_fifos(aec);
+ aec_spk_mic_reset();
+ }
+
+#if DEBUG_AEC
+ /* ref data is 32-bit at this point */
+ size_t ref_bytes = in_frames*aec->num_reference_channels*sizeof(int32_t);
+
+ FILE *fp_in = fopen("/data/local/traces/aec_in.pcm", "a+");
+ if (fp_in) {
+ fwrite((char *)aec->mic_buf, 1, bytes, fp_in);
+ fclose(fp_in);
+ } else {
+ ALOGE("AEC debug: Could not open file aec_in.pcm!");
+ }
+ FILE *fp_out = fopen("/data/local/traces/aec_out.pcm", "a+");
+ if (fp_out) {
+ fwrite((char *)buffer, 1, bytes, fp_out);
+ fclose(fp_out);
+ } else {
+ ALOGE("AEC debug: Could not open file aec_out.pcm!");
+ }
+ FILE *fp_ref = fopen("/data/local/traces/aec_ref.pcm", "a+");
+ if (fp_ref) {
+ fwrite((char *)aec->spk_buf, 1, ref_bytes, fp_ref);
+ fclose(fp_ref);
+ } else {
+ ALOGE("AEC debug: Could not open file aec_ref.pcm!");
+ }
+ FILE *fp_ts = fopen("/data/local/traces/aec_timestamps.txt", "a+");
+ if (fp_ts) {
+ fprintf(fp_ts, "%"PRIu64",%"PRIu64"\n", mic_time, spk_time);
+ fclose(fp_ts);
+ } else {
+ ALOGE("AEC debug: Could not open file aec_timestamps.txt!");
+ }
+#endif /* #if DEBUG_AEC */
+ ALOGV("%s exit", __func__);
+ return ret;
+}
+
+#endif /*#ifdef AEC_HAL*/
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-28 07:03:33 +0000