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NetEq: Deprecate playout modes Fax, Off and Streaming

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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}
This commit is contained in:
Henrik Lundin
2018-06-21 11:13:07 +02:00
committed by Commit Bot
parent c0260b4f2b
commit 80c4cca491
23 changed files with 479 additions and 782 deletions
Download Patch File Download Diff File Expand all files Collapse all files

37
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -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 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
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 @@ TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
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