Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1565133002
Cr-Commit-Position: refs/heads/master@{#11753}
diff --git a/webrtc/api/localaudiosource.cc b/webrtc/api/localaudiosource.cc
index 4b961cf..a118655 100644
--- a/webrtc/api/localaudiosource.cc
+++ b/webrtc/api/localaudiosource.cc
@@ -50,8 +50,7 @@
{MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
{MediaConstraintsInterface::kTypingNoiseDetection,
options->typing_detection},
- {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping},
- {MediaConstraintsInterface::kAecDump, options->aec_dump}
+ {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping}
};
for (const auto& constraint : constraints) {
diff --git a/webrtc/api/localaudiosource_unittest.cc b/webrtc/api/localaudiosource_unittest.cc
index fad78d9..1abb940 100644
--- a/webrtc/api/localaudiosource_unittest.cc
+++ b/webrtc/api/localaudiosource_unittest.cc
@@ -35,7 +35,6 @@
MediaConstraintsInterface::kExperimentalAutoGainControl, true);
constraints.AddMandatory(MediaConstraintsInterface::kNoiseSuppression, false);
constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
- constraints.AddOptional(MediaConstraintsInterface::kAecDump, true);
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
@@ -48,7 +47,6 @@
EXPECT_EQ(rtc::Optional<bool>(true), source->options().experimental_agc);
EXPECT_EQ(rtc::Optional<bool>(false), source->options().noise_suppression);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().highpass_filter);
- EXPECT_EQ(rtc::Optional<bool>(true), source->options().aec_dump);
}
TEST(LocalAudioSourceTest, OptionNotSet) {
diff --git a/webrtc/api/mediaconstraintsinterface.cc b/webrtc/api/mediaconstraintsinterface.cc
index 5152194..b0a68b1 100644
--- a/webrtc/api/mediaconstraintsinterface.cc
+++ b/webrtc/api/mediaconstraintsinterface.cc
@@ -50,7 +50,6 @@
const char MediaConstraintsInterface::kTypingNoiseDetection[] =
"googTypingNoiseDetection";
const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
-const char MediaConstraintsInterface::kAecDump[] = "audioDebugRecording";
// Google-specific constraint keys for a local video source (getUserMedia).
const char MediaConstraintsInterface::kNoiseReduction[] = "googNoiseReduction";
diff --git a/webrtc/api/mediaconstraintsinterface.h b/webrtc/api/mediaconstraintsinterface.h
index ed5d843..0c251f8 100644
--- a/webrtc/api/mediaconstraintsinterface.h
+++ b/webrtc/api/mediaconstraintsinterface.h
@@ -69,7 +69,6 @@
static const char kHighpassFilter[]; // googHighpassFilter
static const char kTypingNoiseDetection[]; // googTypingNoiseDetection
static const char kAudioMirroring[]; // googAudioMirroring
- static const char kAecDump[]; // audioDebugRecording
// Google-specific constraint keys for a local video source
static const char kNoiseReduction[]; // googNoiseReduction
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 8f15878..bb322ed 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -134,7 +134,6 @@
SetFrom(&extended_filter_aec, change.extended_filter_aec);
SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
SetFrom(&experimental_ns, change.experimental_ns);
- SetFrom(&aec_dump, change.aec_dump);
SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
SetFrom(&tx_agc_digital_compression_gain,
change.tx_agc_digital_compression_gain);
@@ -160,7 +159,6 @@
delay_agnostic_aec == o.delay_agnostic_aec &&
experimental_ns == o.experimental_ns &&
adjust_agc_delta == o.adjust_agc_delta &&
- aec_dump == o.aec_dump &&
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
tx_agc_limiter == o.tx_agc_limiter &&
@@ -188,7 +186,6 @@
ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
ost << ToStringIfSet("experimental_ns", experimental_ns);
- ost << ToStringIfSet("aec_dump", aec_dump);
ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
ost << ToStringIfSet("tx_agc_digital_compression_gain",
tx_agc_digital_compression_gain);
@@ -223,7 +220,6 @@
rtc::Optional<bool> extended_filter_aec;
rtc::Optional<bool> delay_agnostic_aec;
rtc::Optional<bool> experimental_ns;
- rtc::Optional<bool> aec_dump;
// Note that tx_agc_* only applies to non-experimental AGC.
rtc::Optional<uint16_t> tx_agc_target_dbov;
rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 70f7f3a..3709e80 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -92,23 +92,6 @@
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
-// Ensure we open the file in a writeable path on ChromeOS and Android. This
-// workaround can be removed when it's possible to specify a filename for audio
-// option based AEC dumps.
-//
-// TODO(grunell): Use a string in the options instead of hardcoding it here
-// and let the embedder choose the filename (crbug.com/264223).
-//
-// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
-// below.
-#if defined(CHROMEOS)
-const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
-#elif defined(ANDROID)
-const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
-#else
-const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
-#endif
-
// Constants from voice_engine_defines.h.
const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
const int kMaxTelephoneEventCode = 255;
@@ -615,7 +598,6 @@
options.extended_filter_aec = rtc::Optional<bool>(false);
options.delay_agnostic_aec = rtc::Optional<bool>(false);
options.experimental_ns = rtc::Optional<bool>(false);
- options.aec_dump = rtc::Optional<bool>(false);
if (!ApplyOptions(options)) {
return false;
}
@@ -868,14 +850,6 @@
}
}
- if (options.aec_dump) {
- LOG(LS_INFO) << "Aec dump is enabled? " << *options.aec_dump;
- if (*options.aec_dump)
- StartAecDump(kAecDumpByAudioOptionFilename);
- else
- StopAecDump();
- }
-
webrtc::Config config;
if (options.delay_agnostic_aec)
