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aec: add new init2 method to initialize with different formats

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The WebRTC echo canceler can support different rates and channels for
the record, out and playback streams.

Add a new method to pass this config to the echo-canceler.
This commit is contained in:
Wim Taymans 2023-04-11 16:54:11 +02:00
parent 8748c77451
commit 45b2983439
2 changed files with 66 additions and 29 deletions
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9
spa/include/spa/interfaces/audio/aec.h
View file

@ -40,7 +40,7 @@ struct spa_audio_aec_events {
};
struct spa_audio_aec_methods {
#define SPA_VERSION_AUDIO_AEC_METHODS 2
#define SPA_VERSION_AUDIO_AEC_METHODS 3
uint32_t version;
int (*add_listener) (void *object,
@ -60,6 +60,12 @@ struct spa_audio_aec_methods {
int (*enum_props) (void* object, int index, struct spa_pod_builder* builder);
int (*get_params) (void* object, struct spa_pod_builder* builder);
int (*set_params) (void *object, const struct spa_pod *args);
/* version 1:3 */
int (*init2) (void *object, const struct spa_dict *args,
struct spa_audio_info_raw *play_info,
struct spa_audio_info_raw *rec_info,
struct spa_audio_info_raw *out_info);
};
#define spa_audio_aec_method(o,method,version,...) \
@ -81,6 +87,7 @@ struct spa_audio_aec_methods {
#define spa_audio_aec_enum_props(o,...) spa_audio_aec_method(o, enum_props, 2, __VA_ARGS__)
#define spa_audio_aec_get_params(o,...) spa_audio_aec_method(o, get_params, 2, __VA_ARGS__)
#define spa_audio_aec_set_params(o,...) spa_audio_aec_method(o, set_params, 2, __VA_ARGS__)
#define spa_audio_aec_init2(o,...) spa_audio_aec_method(o, init2, 3, __VA_ARGS__)
#ifdef __cplusplus
} /* extern "C" */

86
spa/plugins/aec/aec-webrtc.cpp
View file

@ -22,7 +22,9 @@ struct impl_data {
struct spa_log *log;
std::unique_ptr<webrtc::AudioProcessing> apm;
spa_audio_info_raw info;
spa_audio_info_raw rec_info;
spa_audio_info_raw out_info;
spa_audio_info_raw play_info;
std::unique_ptr<float *[]> play_buffer, rec_buffer, out_buffer;
};
@ -38,9 +40,12 @@ static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bo
return default_value;
}
static int webrtc_init(void *object, const struct spa_dict *args, const struct spa_audio_info_raw *info)
static int webrtc_init2(void *object, const struct spa_dict *args,
struct spa_audio_info_raw *rec_info, struct spa_audio_info_raw *out_info,
struct spa_audio_info_raw *play_info)
{
auto impl = static_cast<struct impl_data*>(object);
int res;
bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
@ -67,16 +72,16 @@ static int webrtc_init(void *object, const struct spa_dict *args, const struct s
config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(experimental_ns));
webrtc::ProcessingConfig pconfig = {{
webrtc::StreamConfig(info->rate, info->channels, false), /* input stream */
webrtc::StreamConfig(info->rate, info->channels, false), /* output stream */
webrtc::StreamConfig(info->rate, info->channels, false), /* reverse input stream */
webrtc::StreamConfig(info->rate, info->channels, false), /* reverse output stream */
webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
webrtc::StreamConfig(out_info->rate, out_info->channels, false), /* output stream */
webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse input stream */
webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
}};
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
spa_log_error(impl->log, "Error initialising webrtc audio processing module");
return -1;
if ((res = apm->Initialize(pconfig)) != webrtc::AudioProcessing::kNoError) {
spa_log_error(impl->log, "Error initialising webrtc audio processing module: %d", res);
return -EINVAL;
}
apm->high_pass_filter()->Enable(high_pass_filter);
@ -94,48 +99,72 @@ static int webrtc_init(void *object, const struct spa_dict *args, const struct s
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
apm->gain_control()->Enable(gain_control);
impl->apm = std::move(apm);
impl->info = *info;
impl->play_buffer = std::make_unique<float *[]>(info->channels);
impl->rec_buffer = std::make_unique<float *[]>(info->channels);
impl->out_buffer = std::make_unique<float *[]>(info->channels);
impl->rec_info = *rec_info;
impl->out_info = *out_info;
impl->play_info = *play_info;
impl->play_buffer = std::make_unique<float *[]>(play_info->channels);
impl->rec_buffer = std::make_unique<float *[]>(rec_info->channels);
impl->out_buffer = std::make_unique<float *[]>(out_info->channels);
return 0;
}
static int webrtc_init(void *object, const struct spa_dict *args,
const struct spa_audio_info_raw *info)
{
int res;
struct spa_audio_info_raw rec_info = *info;
struct spa_audio_info_raw out_info = *info;
struct spa_audio_info_raw play_info = *info;
res = webrtc_init2(object, args, &rec_info, &out_info, &play_info);
if (rec_info.channels != out_info.channels)
res = -EINVAL;
return res;
}
static int webrtc_run(void *object, const float *rec[], const float *play[], float *out[], uint32_t n_samples)
{
auto impl = static_cast<struct impl_data*>(object);
webrtc::StreamConfig config =
webrtc::StreamConfig(impl->info.rate, impl->info.channels, false);
unsigned int num_blocks = n_samples * 1000 / impl->info.rate / 10;
int res;
if (n_samples * 1000 / impl->info.rate % 10 != 0) {
webrtc::StreamConfig play_config =
webrtc::StreamConfig(impl->play_info.rate, impl->play_info.channels, false);
webrtc::StreamConfig rec_config =
webrtc::StreamConfig(impl->rec_info.rate, impl->rec_info.channels, false);
webrtc::StreamConfig out_config =
webrtc::StreamConfig(impl->out_info.rate, impl->out_info.channels, false);
unsigned int num_blocks = n_samples * 1000 / impl->play_info.rate / 10;
if (n_samples * 1000 / impl->play_info.rate % 10 != 0) {
spa_log_error(impl->log, "Buffers must be multiples of 10ms in length (currently %u samples)", n_samples);
return -1;
return -EINVAL;
}
for (size_t i = 0; i < num_blocks; i ++) {
for (size_t j = 0; j < impl->info.channels; j++) {
impl->play_buffer[j] = const_cast<float *>(play[j]) + config.num_frames() * i;
impl->rec_buffer[j] = const_cast<float *>(rec[j]) + config.num_frames() * i;
impl->out_buffer[j] = out[j] + config.num_frames() * i;
}
for (size_t j = 0; j < impl->play_info.channels; j++)
impl->play_buffer[j] = const_cast<float *>(play[j]) + play_config.num_frames() * i;
for (size_t j = 0; j < impl->rec_info.channels; j++)
impl->rec_buffer[j] = const_cast<float *>(rec[j]) + rec_config.num_frames() * i;
for (size_t j = 0; j < impl->out_info.channels; j++)
impl->out_buffer[j] = out[j] + out_config.num_frames() * i;
/* FIXME: ProcessReverseStream may change the playback buffer, in which
* case we should use that, if we ever expose the intelligibility
* enhancer */
if (impl->apm->ProcessReverseStream(impl->play_buffer.get(), config, config, impl->play_buffer.get()) !=
if ((res = impl->apm->ProcessReverseStream(impl->play_buffer.get(),
play_config, play_config, impl->play_buffer.get())) !=
webrtc::AudioProcessing::kNoError) {
spa_log_error(impl->log, "Processing reverse stream failed");
spa_log_error(impl->log, "Processing reverse stream failed: %d", res);
}
// Extra delay introduced by multiple frames
impl->apm->set_stream_delay_ms((num_blocks - 1) * 10);
if (impl->apm->ProcessStream(impl->rec_buffer.get(), config, config, impl->out_buffer.get()) !=
if ((res = impl->apm->ProcessStream(impl->rec_buffer.get(),
rec_config, out_config, impl->out_buffer.get())) !=
webrtc::AudioProcessing::kNoError) {
spa_log_error(impl->log, "Processing stream failed");
spa_log_error(impl->log, "Processing stream failed: %d", res);
}
}
return 0;
}
@ -144,6 +173,7 @@ static const struct spa_audio_aec_methods impl_aec = {
.add_listener = NULL,
.init = webrtc_init,
.run = webrtc_run,
.init2 = webrtc_init2,
};
static int impl_get_interface(struct spa_handle *handle, const char *type, void **interface)