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echo-cancel: Update webrtc-audio-processing usage to new API

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The code now needs C++11 support to compile with the updated
webrtc-audio-processing library.
This commit is contained in:
Arun Raghavan 2016-02-17 19:46:53 +05:30 • committed by Tanu Kaskinen
parent 93822f98f4
commit f8beaae238
5 changed files with 630 additions and 27 deletions
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54
src/modules/echo-cancel/webrtc.cc
View file

@ -33,8 +33,8 @@ PA_C_DECL_BEGIN
#include "echo-cancel.h"
PA_C_DECL_END
#include <audio_processing.h>
#include <module_common_types.h>
#include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#define BLOCK_SIZE_US 10000
@ -80,6 +80,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
pa_sample_spec *out_ss, pa_channel_map *out_map,
uint32_t *nframes, const char *args) {
webrtc::AudioProcessing *apm = NULL;
webrtc::ProcessingConfig pconfig;
bool hpf, ns, agc, dgc, mobile, cn;
int rm = -1;
pa_modargs *ma;
@ -153,7 +154,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
}
}
apm = webrtc::AudioProcessing::Create(0);
apm = webrtc::AudioProcessing::Create();
out_ss->format = PA_SAMPLE_S16NE;
*play_ss = *out_ss;
@ -163,22 +164,19 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
*rec_ss = *out_ss;
*rec_map = *out_map;
apm->set_sample_rate_hz(out_ss->rate);
apm->set_num_channels(out_ss->channels, out_ss->channels);
apm->set_num_reverse_channels(play_ss->channels);
pconfig = {
webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */
webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* output stream */
webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */
webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */
};
apm->Initialize(pconfig);
if (hpf)
apm->high_pass_filter()->Enable(true);
if (!mobile) {
if (ec->params.drift_compensation) {
apm->echo_cancellation()->set_device_sample_rate_hz(out_ss->rate);
apm->echo_cancellation()->enable_drift_compensation(true);
} else {
apm->echo_cancellation()->enable_drift_compensation(false);
}
apm->echo_cancellation()->enable_drift_compensation(ec->params.drift_compensation);
apm->echo_cancellation()->Enable(true);
} else {
apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
@ -225,7 +223,7 @@ fail:
if (ma)
pa_modargs_free(ma);
if (apm)
webrtc::AudioProcessing::Destroy(apm);
delete apm;
return false;
}
@ -235,10 +233,13 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioFrame play_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
play_frame._audioChannel = ss->channels;
play_frame._frequencyInHz = ss->rate;
play_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(play_frame._payloadData, play, ec->params.priv.webrtc.blocksize);
play_frame.num_channels_ = ss->channels;
play_frame.sample_rate_hz_ = ss->rate;
play_frame.interleaved_ = true;
play_frame.samples_per_channel_ = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
memcpy(play_frame.data_, play, ec->params.priv.webrtc.blocksize);
apm->AnalyzeReverseStream(&play_frame);
}
@ -249,10 +250,13 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
pa_cvolume v;
out_frame._audioChannel = ss->channels;
out_frame._frequencyInHz = ss->rate;
out_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(out_frame._payloadData, rec, ec->params.priv.webrtc.blocksize);
out_frame.num_channels_ = ss->channels;
out_frame.sample_rate_hz_ = ss->rate;
out_frame.interleaved_ = true;
out_frame.samples_per_channel_ = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
memcpy(out_frame.data_, rec, ec->params.priv.webrtc.blocksize);
if (ec->params.priv.webrtc.agc) {
pa_cvolume_init(&v);
@ -268,7 +272,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
pa_echo_canceller_set_capture_volume(ec, &v);
}
memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
memcpy(out, out_frame.data_, ec->params.priv.webrtc.blocksize);
}
void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
@ -285,7 +289,7 @@ void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *
void pa_webrtc_ec_done(pa_echo_canceller *ec) {
if (ec->params.priv.webrtc.apm) {
webrtc::AudioProcessing::Destroy((webrtc::AudioProcessing*)ec->params.priv.webrtc.apm);
delete (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
ec->params.priv.webrtc.apm = NULL;
}
}