From: Arun Raghavan <git@xxxxxxxxxxxxxxxx>
---
configure.ac | 2 +-
src/Makefile.am | 2 +-
src/modules/echo-cancel/webrtc.cc | 54 +++++++++++++++++++++------------------
3 files changed, 31 insertions(+), 27 deletions(-)
diff --git a/configure.ac b/configure.ac
index 003673e..a9b2bbb 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1370,7 +1370,7 @@ AC_ARG_ENABLE([webrtc-aec],
AS_HELP_STRING([--enable-webrtc-aec], [Enable the optional WebRTC-based echo canceller]))
AS_IF([test "x$enable_webrtc_aec" != "xno"],
- [PKG_CHECK_MODULES(WEBRTC, [ webrtc-audio-processing ], [HAVE_WEBRTC=1], [HAVE_WEBRTC=0])],
+ [PKG_CHECK_MODULES(WEBRTC, [ webrtc-audio-processing > 0.1 ], [HAVE_WEBRTC=1], [HAVE_WEBRTC=0])],
[HAVE_WEBRTC=0])
AS_IF([test "x$enable_webrtc_aec" = "xyes" && test "x$HAVE_WEBRTC" = "x0"],
diff --git a/src/Makefile.am b/src/Makefile.am
index 0c58d37..d3b384c 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -49,7 +49,7 @@ AM_CPPFLAGS = \
-DPA_BUILDDIR=\"$(abs_builddir)\"
AM_CFLAGS = \
$(PTHREAD_CFLAGS)
-AM_CXXFLAGS = $(AM_CFLAGS)
+AM_CXXFLAGS = $(AM_CFLAGS) -std=c++11
SERVER_CFLAGS = -D__INCLUDED_FROM_PULSE_AUDIO
AM_LIBADD = $(PTHREAD_LIBS) $(INTLLIBS)
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 511c7ee..4a23377 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -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(out_ss->rate, out_ss->channels, false), /* input stream */
+ webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* output stream */
+ webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* reverse input stream */
+ webrtc::StreamConfig(out_ss->rate, out_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_ = false;
+ 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_ = false;
+ 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;
}
}
--
2.4.3
