NetEq: Deprecate playout modes Fax, Off and Streaming
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 4ed7a6b..27e9535 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -25,6 +25,8 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/messagedigest.h"
@@ -555,7 +557,7 @@
class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
protected:
NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
- config_.playout_mode = kPlayoutFax;
+ config_.for_test_no_time_stretching = true;
}
void TestJitterBufferDelay(bool apply_packet_loss);
};
@@ -1723,4 +1725,37 @@
TestJitterBufferDelay(true);
}
+namespace test {
+// TODO(henrik.lundin) NetEqRtpDumpInput requires protobuf support. It shouldn't
+// need it, but because it is bundled with NetEqEventLogInput, it is neded.
+// This should be refactored.
+#if WEBRTC_ENABLE_PROTOBUF
+TEST(NetEqNoTimeStretchingMode, RunTest) {
+ NetEq::Config config;
+ config.for_test_no_time_stretching = true;
+ auto codecs = NetEqTest::StandardDecoderMap();
+ NetEqTest::ExtDecoderMap ext_codecs;
+ NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
+ {1, kRtpExtensionAudioLevel},
+ {3, kRtpExtensionAbsoluteSendTime},
+ {5, kRtpExtensionTransportSequenceNumber},
+ {7, kRtpExtensionVideoContentType},
+ {8, kRtpExtensionVideoTiming}};
+ std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
+ webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
+ rtp_ext_map));
+ std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
+ new TimeLimitedNetEqInput(std::move(input), 20000));
+ std::unique_ptr<AudioSink> output(new VoidAudioSink);
+ NetEqTest::Callbacks callbacks;
+ NetEqTest test(config, codecs, ext_codecs, std::move(input_time_limit),
+ std::move(output), callbacks);
+ test.Run();
+ const auto stats = test.SimulationStats();
+ EXPECT_EQ(0, stats.accelerate_rate);
+ EXPECT_EQ(0, stats.preemptive_rate);
+}
+#endif
+
+} // namespace test
} // namespace webrtc