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pipewire: module-echo-cancel: aec-webrtc: use unique_ptr

Browse source 
Use `std::unique_ptr<>` to clean up memory in the destructor
instead of manually freeing resources in the destroy callback
and error code paths.
This commit is contained in:
Barnabás Pőcze 2021-11-09 18:04:09 +01:00
parent 5896083545
commit 336e4d5f03
1 changed files with 21 additions and 36 deletions
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57
src/modules/module-echo-cancel/aec-webrtc.cpp
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@ -23,6 +23,9 @@
* DEALINGS IN THE SOFTWARE.
*/
#include <memory>
#include <utility>
#include "echo-cancel.h"
#include <pipewire/pipewire.h>
@ -32,20 +35,21 @@
#include <webrtc/system_wrappers/include/trace.h>
struct impl {
webrtc::AudioProcessing *apm = NULL;
std::unique_ptr<webrtc::AudioProcessing> apm;
spa_audio_info_raw info;
float** play_buffer;
float** rec_buffer;
float** out_buffer;
std::unique_ptr<float *[]> play_buffer, rec_buffer, out_buffer;
impl(std::unique_ptr<webrtc::AudioProcessing> apm, const spa_audio_info_raw& info)
: apm(std::move(apm)),
info(info),
play_buffer(std::make_unique<float *[]>(info.channels)),
rec_buffer(std::make_unique<float *[]>(info.channels)),
out_buffer(std::make_unique<float *[]>(info.channels))
{ }
};
static void *webrtc_create(const struct pw_properties *args, const spa_audio_info_raw *info)
{
struct impl *impl;
webrtc::AudioProcessing *apm;
webrtc::ProcessingConfig pconfig;
webrtc::Config config;
bool extended_filter = pw_properties_get_bool(args, "webrtc.extended_filter", true);
bool delay_agnostic = pw_properties_get_bool(args, "webrtc.delay_agnostic", true);
bool high_pass_filter = pw_properties_get_bool(args, "webrtc.high_pass_filter", true);
@ -63,24 +67,24 @@ static void *webrtc_create(const struct pw_properties *args, const spa_audio_inf
// Disable by default
bool intelligibility = pw_properties_get_bool(args, "webrtc.intelligibility", false);
webrtc::Config config;
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(experimental_agc));
config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(experimental_ns));
config.Set<webrtc::Intelligibility>(new webrtc::Intelligibility(intelligibility));
apm = webrtc::AudioProcessing::Create(config);
pconfig = {{
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 */
}};
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
pw_log_error("Error initialising webrtc audio processing module");
goto error;
return nullptr;
}
apm->high_pass_filter()->Enable(high_pass_filter);
@ -96,33 +100,14 @@ static void *webrtc_create(const struct pw_properties *args, const spa_audio_inf
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
apm->gain_control()->Enable(gain_control);
impl = (struct impl *)calloc(1, sizeof(struct impl));
impl->info = *info;
impl->play_buffer = (float **)calloc(info->channels, sizeof(float*));
impl->rec_buffer = (float **)calloc(info->channels, sizeof(float*));
impl->out_buffer = (float **)calloc(info->channels, sizeof(float*));
impl->apm = apm;
return impl;
error:
if (apm)
delete apm;
return NULL;
return new impl(std::move(apm), *info);
}
static void webrtc_destroy(void *ec)
{
auto impl = static_cast<struct impl *>(ec);
delete impl->apm;
free(impl->play_buffer);
free(impl->rec_buffer);
free(impl->out_buffer);
free(impl);
delete impl;
}
static int webrtc_run(void *ec, const float *rec[], const float *play[], float *out[], uint32_t n_samples)
@ -146,7 +131,7 @@ static int webrtc_run(void *ec, const float *rec[], const float *play[], float *
/* 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, config, config, impl->play_buffer) !=
if (impl->apm->ProcessReverseStream(impl->play_buffer.get(), config, config, impl->play_buffer.get()) !=
webrtc::AudioProcessing::kNoError) {
pw_log_error("Processing reverse stream failed");
}
@ -154,7 +139,7 @@ static int webrtc_run(void *ec, const float *rec[], const float *play[], float *
// Extra delay introduced by multiple frames
impl->apm->set_stream_delay_ms((num_blocks - 1) * 10);
if (impl->apm->ProcessStream(impl->rec_buffer, config, config, impl->out_buffer) !=
if (impl->apm->ProcessStream(impl->rec_buffer.get(), config, config, impl->out_buffer.get()) !=
webrtc::AudioProcessing::kNoError) {
pw_log_error("Processing stream failed");
}