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Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"

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This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1

Original change's description:
> 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}

Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
This commit is contained in:
Henrik Lundin
2018-07-02 10:14:46 +02:00
committed by Commit Bot
parent 04b18cb365
commit 7687ad58b2
23 changed files with 483 additions and 782 deletions
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37
modules/audio_coding/neteq/neteq_unittest.cc
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@ -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