Remove all use of AcmReceiver from WebRTC
The class itself and its unit test remains, for now, but will be removed
later.
Bug: webrtc:14867
Change-Id: I36cec8fca7913663f63c53622ed2760e5e048c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362580
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43023}
diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index 42fc665..2ae743a 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -64,8 +64,10 @@
"../../api/audio:audio_mixer_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/environment",
+ "../../api/neteq:neteq_api",
"../../api/voip:voip_api",
"../../modules/audio_coding",
+ "../../modules/audio_coding:default_neteq_factory",
"../../modules/rtp_rtcp",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:criticalsection",
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 50332c1..82b4012 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_format.h"
#include "audio/utility/audio_frame_operations.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
@@ -30,12 +31,10 @@
namespace {
-acm2::AcmReceiver::Config CreateAcmConfig(
- rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
- acm2::AcmReceiver::Config acm_config;
- acm_config.neteq_config.enable_muted_state = true;
- acm_config.decoder_factory = decoder_factory;
- return acm_config;
+NetEq::Config CreateNetEqConfig() {
+ NetEq::Config config;
+ config.enable_muted_state = true;
+ return config;
}
} // namespace
@@ -51,7 +50,9 @@
first_rtp_timestamp_(-1),
rtp_receive_statistics_(receive_statistics),
rtp_rtcp_(rtp_rtcp),
- acm_receiver_(env_, CreateAcmConfig(decoder_factory)),
+ neteq_(DefaultNetEqFactory().Create(env,
+ CreateNetEqConfig(),
+ decoder_factory)),
ntp_estimator_(&env_.clock()) {}
AudioIngress::~AudioIngress() = default;
@@ -63,13 +64,17 @@
// Get 10ms raw PCM data from the ACM.
bool muted = false;
- if (acm_receiver_.GetAudio(sampling_rate, audio_frame, &muted) == -1) {
- RTC_DLOG(LS_ERROR) << "GetAudio() failed!";
- // In all likelihood, the audio in this frame is garbage. We return an
- // error so that the audio mixer module doesn't add it to the mix. As
- // a result, it won't be played out and the actions skipped here are
- // irrelevant.
- return AudioMixer::Source::AudioFrameInfo::kError;
+ {
+ MutexLock lock(&lock_);
+ if ((neteq_->GetAudio(audio_frame, &muted) != NetEq::kOK) ||
+ !resampler_helper_.MaybeResample(sampling_rate, audio_frame)) {
+ RTC_DLOG(LS_ERROR) << "GetAudio() failed!";
+ // In all likelihood, the audio in this frame is garbage. We return an
+ // error so that the audio mixer module doesn't add it to the mix. As
+ // a result, it won't be played out and the actions skipped here are
+ // irrelevant.
+ return AudioMixer::Source::AudioFrameInfo::kError;
+ }
}
if (muted) {
@@ -103,10 +108,10 @@
}
// For clock rate, default to the playout sampling rate if we haven't
// received any packets yet.
- std::optional<std::pair<int, SdpAudioFormat>> decoder =
- acm_receiver_.LastDecoder();
- int clock_rate = decoder ? decoder->second.clockrate_hz
- : acm_receiver_.last_output_sample_rate_hz();
+ std::optional<NetEq::DecoderFormat> decoder =
+ neteq_->GetCurrentDecoderFormat();
+ int clock_rate = decoder ? decoder->sdp_format.clockrate_hz
+ : neteq_->last_output_sample_rate_hz();
RTC_DCHECK_GT(clock_rate, 0);
audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - first_rtp_timestamp_) / (clock_rate / 1000);
@@ -136,7 +141,7 @@
receive_codec_info_[kv.first] = kv.second.clockrate_hz;
}
}
- acm_receiver_.SetCodecs(codecs);
+ neteq_->SetCodecs(codecs);
}
void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
@@ -186,10 +191,12 @@
auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
// Push the incoming payload (parsed and ready for decoding) into the ACM.
- if (acm_receiver_.InsertPacket(header, data_view,
- env_.clock().CurrentTime()) != 0) {
- RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
- "push data to the ACM";
+ if (data_view.empty()) {
+ neteq_->InsertEmptyPacket(header);
+ } else if (neteq_->InsertPacket(header, data_view,
+ env_.clock().CurrentTime()) < 0) {
+ RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
+ "insert packet into NetEq";
}
}
@@ -235,16 +242,63 @@
}
}
+NetworkStatistics AudioIngress::GetNetworkStatistics() const {
+ NetworkStatistics stats;
+ stats.currentExpandRate = 0;
+ stats.currentSpeechExpandRate = 0;
+ stats.currentPreemptiveRate = 0;
+ stats.currentAccelerateRate = 0;
+ stats.currentSecondaryDecodedRate = 0;
+ stats.currentSecondaryDiscardedRate = 0;
+ stats.meanWaitingTimeMs = -1;
+ stats.maxWaitingTimeMs = 1;
+
+ NetEqNetworkStatistics neteq_stat = neteq_->CurrentNetworkStatistics();
+ stats.currentBufferSize = neteq_stat.current_buffer_size_ms;
+ stats.preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
+ stats.jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
+
+ NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
+ stats.totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
+ stats.concealedSamples = neteq_lifetime_stat.concealed_samples;
+ stats.silentConcealedSamples = neteq_lifetime_stat.silent_concealed_samples;
+ stats.concealmentEvents = neteq_lifetime_stat.concealment_events;
+ stats.jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+ stats.jitterBufferTargetDelayMs =
+ neteq_lifetime_stat.jitter_buffer_target_delay_ms;
+ stats.jitterBufferMinimumDelayMs =
+ neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
+ stats.jitterBufferEmittedCount =
+ neteq_lifetime_stat.jitter_buffer_emitted_count;
+ stats.delayedPacketOutageSamples =
+ neteq_lifetime_stat.delayed_packet_outage_samples;
+ stats.relativePacketArrivalDelayMs =
+ neteq_lifetime_stat.relative_packet_arrival_delay_ms;
+ stats.interruptionCount = neteq_lifetime_stat.interruption_count;
+ stats.totalInterruptionDurationMs =
+ neteq_lifetime_stat.total_interruption_duration_ms;
+ stats.insertedSamplesForDeceleration =
+ neteq_lifetime_stat.inserted_samples_for_deceleration;
+ stats.removedSamplesForAcceleration =
+ neteq_lifetime_stat.removed_samples_for_acceleration;
+ stats.fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
+ stats.fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
+ stats.totalProcessingDelayUs = neteq_lifetime_stat.total_processing_delay_us;
+ stats.packetsDiscarded = neteq_lifetime_stat.packets_discarded;
+
+ NetEqOperationsAndState neteq_operations_and_state =
+ neteq_->GetOperationsAndState();
+ stats.packetBufferFlushes = neteq_operations_and_state.packet_buffer_flushes;
+
+ return stats;
+}
+
ChannelStatistics AudioIngress::GetChannelStatistics() {
ChannelStatistics channel_stats;
// Get clockrate for current decoder ahead of jitter calculation.
- uint32_t clockrate_hz = 0;
- std::optional<std::pair<int, SdpAudioFormat>> decoder =
- acm_receiver_.LastDecoder();
- if (decoder) {
- clockrate_hz = decoder->second.clockrate_hz;
- }
+ auto decoder = neteq_->GetCurrentDecoderFormat();
+ const uint32_t clockrate_hz = decoder ? decoder->sdp_format.clockrate_hz : 0;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(remote_ssrc_);
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index 9651871..3569f21 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -20,12 +20,14 @@
#include "api/array_view.h"
#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/environment/environment.h"
+#include "api/neteq/neteq.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
#include "api/voip/voip_statistics.h"
#include "audio/audio_level.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
@@ -81,12 +83,7 @@
return output_audio_level_.TotalDuration();
}
- NetworkStatistics GetNetworkStatistics() const {
- NetworkStatistics stats;
- acm_receiver_.GetNetworkStatistics(&stats,
- /*get_and_clear_legacy_stats=*/false);
- return stats;
- }
+ NetworkStatistics GetNetworkStatistics() const;
ChannelStatistics GetChannelStatistics();
@@ -98,11 +95,14 @@
return rtc::dchecked_cast<int>(remote_ssrc_.load());
}
int PreferredSampleRate() const override {
+ std::optional<NetEq::DecoderFormat> decoder =
+ neteq_->GetCurrentDecoderFormat();
+
// If we haven't received any RTP packet from remote and thus
// last_packet_sampling_rate is not available then use NetEq's sampling
// rate as that would be what would be used for audio output sample.
- return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
- acm_receiver_.last_output_sample_rate_hz());
+ return std::max(decoder ? decoder->sample_rate_hz : 0,
+ neteq_->last_output_sample_rate_hz());
}
private:
@@ -126,8 +126,8 @@
// Synchronizaton is handled internally by RtpRtcpInterface.
RtpRtcpInterface* const rtp_rtcp_;
- // Synchronizaton is handled internally by acm2::AcmReceiver.
- acm2::AcmReceiver acm_receiver_;
+ // Synchronizaton is handled internally by NetEq.
+ const std::unique_ptr<NetEq> neteq_;
// Synchronizaton is handled internally by voe::AudioLevel.
voe::AudioLevel output_audio_level_;
@@ -141,6 +141,9 @@
std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
+
+ // Resampler for the output audio.
+ acm2::ResamplerHelper resampler_helper_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index c92a26f..8c42487 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1067,8 +1067,10 @@
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../api/environment",
"../../api/environment:environment_factory",
+ "../../api/neteq:neteq_api",
"../../api/units:timestamp",
"../../common_audio",
+ "../../modules/audio_coding:default_neteq_factory",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
@@ -1126,6 +1128,8 @@
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/environment:environment_factory",
+ "../../api/neteq:neteq_api",
+ "../../modules/audio_coding:default_neteq_factory",
"../../test:test_support",
"//testing/gtest",
]
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index fa48433..e23dfc7 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -17,6 +17,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
@@ -25,15 +26,6 @@
namespace webrtc {
namespace test {
-namespace {
-acm2::AcmReceiver::Config MakeAcmConfig(
- rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
- acm2::AcmReceiver::Config config;
- config.decoder_factory = std::move(decoder_factory);
- return config;
-}
-} // namespace
-
AcmReceiveTestOldApi::AcmReceiveTestOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
@@ -41,9 +33,9 @@
NumOutputChannels exptected_output_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: clock_(0),
- acm_receiver_(std::make_unique<acm2::AcmReceiver>(
- CreateEnvironment(&clock_),
- MakeAcmConfig(std::move(decoder_factory)))),
+ neteq_(DefaultNetEqFactory().Create(CreateEnvironment(&clock_),
+ NetEq::Config(),
+ std::move(decoder_factory))),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
@@ -52,43 +44,43 @@
AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
- acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ neteq_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp.
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
- acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {93, {"L16", 8000, 1}},
- {94, {"L16", 16000, 1}},
- {95, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {120, {"OPUS", 48000, 2}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ neteq_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {93, {"L16", 8000, 1}},
+ {94, {"L16", 16000, 1}},
+ {95, {"L16", 32000, 1}},
+ {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {120, {"OPUS", 48000, 2}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
void AcmReceiveTestOldApi::Run() {
@@ -98,8 +90,9 @@
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
bool muted;
- EXPECT_EQ(
- 0, acm_receiver_->GetAudio(output_freq_hz_, &output_frame, &muted));
+ EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_frame, &muted));
+ EXPECT_TRUE(
+ resampler_helper_.MaybeResample(output_freq_hz_, &output_frame));
ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
ASSERT_FALSE(muted);
const size_t samples_per_block =
@@ -119,7 +112,7 @@
AfterGetAudio();
}
- EXPECT_EQ(0, acm_receiver_->InsertPacket(
+ EXPECT_EQ(0, neteq_->InsertPacket(
packet->header(),
rtc::ArrayView<const uint8_t>(
packet->payload(), packet->payload_length_bytes()),
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index d0195dd..5e775af 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -17,8 +17,9 @@
#include <string>
#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
#include "api/scoped_refptr.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -63,7 +64,8 @@
virtual void AfterGetAudio() {}
SimulatedClock clock_;
- std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
+ std::unique_ptr<NetEq> neteq_;
+ acm2::ResamplerHelper resampler_helper_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index c7e9dfc..62c3c39 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -34,6 +34,7 @@
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
@@ -174,9 +175,8 @@
void SetUp() {
acm_ = AudioCodingModule::Create();
- acm2::AcmReceiver::Config config;
- config.decoder_factory = CreateBuiltinAudioDecoderFactory();
- acm_receiver_ = std::make_unique<acm2::AcmReceiver>(env_, config);
+ neteq_ = DefaultNetEqFactory().Create(env_, NetEq::Config(),
+ CreateBuiltinAudioDecoderFactory());
rtp_utility_->Populate(&rtp_header_);
@@ -199,7 +199,7 @@
}
virtual void RegisterCodec() {
- acm_receiver_->SetCodecs({{kPayloadType, *audio_format_}});
+ neteq_->SetCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->Create(
env_, *audio_format_, {.payload_type = kPayloadType}));
}
@@ -211,7 +211,7 @@
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
- ASSERT_EQ(0, acm_receiver_->InsertPacket(
+ ASSERT_EQ(0, neteq_->InsertPacket(
rtp_header_,
rtc::ArrayView<const uint8_t>(kPayload, kPayloadSizeBytes),
/*receive_time=*/Timestamp::MinusInfinity()));
@@ -221,7 +221,7 @@
virtual void PullAudio() {
AudioFrame audio_frame;
bool muted;
- ASSERT_EQ(0, acm_receiver_->GetAudio(-1, &audio_frame, &muted));
+ ASSERT_EQ(0, neteq_->GetAudio(&audio_frame, &muted));
ASSERT_FALSE(muted);
}
@@ -244,7 +244,7 @@
Environment env_;
std::unique_ptr<RtpData> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
- std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
+ std::unique_ptr<NetEq> neteq_;
PacketizationCallbackStubOldApi packet_cb_;
RTPHeader rtp_header_;
AudioFrame input_frame_;
@@ -256,19 +256,6 @@
class AudioCodingModuleTestOldApiDeathTest
: public AudioCodingModuleTestOldApi {};
-// The below test is temporarily disabled on Windows due to problems
-// with clang debug builds.
-// TODO(tommi): Re-enable when we've figured out what the problem is.
-// http://crbug.com/615050
-#if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \
- GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(AudioCodingModuleTestOldApiDeathTest, FailOnZeroDesiredFrequency) {
- AudioFrame audio_frame;
- bool muted;
- RTC_EXPECT_DEATH(acm_receiver_->GetAudio(0, &audio_frame, &muted), "");
-}
-#endif
-
// Checks that the transport callback is invoked once for each speech packet.
// Also checks that the frame type is kAudioFrameSpeech.
TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
@@ -297,8 +284,8 @@
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
- acm_receiver_->SetCodecs({{kPayloadType, *audio_format_},
- {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
+ neteq_->SetCodecs({{kPayloadType, *audio_format_},
+ {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(*enc);
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index f09b357..705f8a1 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -82,7 +82,7 @@
return 0;
}
- status = _receiverACM->InsertPacket(
+ status = _neteq->InsertPacket(
rtp_header, rtc::ArrayView<const uint8_t>(_payloadData, payloadDataSize),
/*receive_time=*/Timestamp::MinusInfinity());
@@ -187,7 +187,7 @@
}
Channel::Channel(int16_t chID)
- : _receiverACM(NULL),
+ : _neteq(NULL),
_seqNo(0),
_bitStreamFile(NULL),
_saveBitStream(false),
@@ -229,8 +229,8 @@
Channel::~Channel() {}
-void Channel::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
- _receiverACM = acm_receiver;
+void Channel::RegisterReceiverNetEq(NetEq* neteq) {
+ _neteq = neteq;
return;
}
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index ebf4461..616e283 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -13,7 +13,7 @@
#include <stdio.h>
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/synchronization/mutex.h"
@@ -55,7 +55,7 @@
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) override;
- void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
+ void RegisterReceiverNetEq(NetEq* neteq);
void ResetStats();
@@ -84,7 +84,7 @@
private:
void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
- acm2::AcmReceiver* _receiverACM;
+ NetEq* _neteq;
uint16_t _seqNo;
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
uint8_t _payloadData[60 * 32 * 2 * 2];
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 70e18b3..665d0d9 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -21,6 +21,7 @@
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -105,32 +106,32 @@
: _playoutLengthSmpls(kWebRtc10MsPcmAudio),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
-void Receiver::Setup(acm2::AcmReceiver* acm_receiver,
+void Receiver::Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
int file_num) {
if (channels == 1) {
- acm_receiver->SetCodecs({{107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {120, {"OPUS", 48000, 2}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ neteq->SetCodecs({{107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {120, {"OPUS", 48000, 2}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
} else {
ASSERT_EQ(channels, 2u);
- acm_receiver->SetCodecs({{111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {110, {"PCMU", 8000, 2}},
- {118, {"PCMA", 8000, 2}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ neteq->SetCodecs({{111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {110, {"PCMU", 8000, 2}},
+ {118, {"PCMA", 8000, 2}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
}
int playSampFreq;
@@ -147,7 +148,7 @@
_realPayloadSizeBytes = 0;
_playoutBuffer = new int16_t[kWebRtc10MsPcmAudio];
_frequency = playSampFreq;
- _acm_receiver = acm_receiver;
+ _neteq = neteq;
_firstTime = true;
}
@@ -172,11 +173,11 @@
}
}
- EXPECT_EQ(0, _acm_receiver->InsertPacket(
- _rtpHeader,
- rtc::ArrayView<const uint8_t>(_incomingPayload,
- _realPayloadSizeBytes),
- /*receive_time=*/Timestamp::Millis(_nextTime)));
+ EXPECT_GE(
+ 0, _neteq->InsertPacket(_rtpHeader,
+ rtc::ArrayView<const uint8_t>(
+ _incomingPayload, _realPayloadSizeBytes),
+ /*receive_time=*/Timestamp::Millis(_nextTime)));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
@@ -189,18 +190,19 @@
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
bool muted;
- int32_t ok = _acm_receiver->GetAudio(_frequency, &audioFrame, &muted);
+ int ok = _neteq->GetAudio(&audioFrame, &muted);
if (muted) {
ADD_FAILURE();
return false;
}
- EXPECT_EQ(0, ok);
+ EXPECT_EQ(NetEq::kOK, ok);
if (ok < 0) {
return false;
}
if (_playoutLengthSmpls == 0) {
return false;
}
+ EXPECT_TRUE(_resampler_helper.MaybeResample(_frequency, &audioFrame));
_pcmFile.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_ *
audioFrame.num_channels_);
return true;
@@ -264,12 +266,10 @@
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
- std::unique_ptr<acm2::AcmReceiver> acm_receiver =
- std::make_unique<acm2::AcmReceiver>(
- env, acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()));
+ std::unique_ptr<NetEq> neteq = DefaultNetEqFactory().Create(
+ env, NetEq::Config(), CreateBuiltinAudioDecoderFactory());
Receiver receiver;
- receiver.Setup(acm_receiver.get(), &rtpFile, "encodeDecode_out", 1,
- file_num);
+ receiver.Setup(neteq.get(), &rtpFile, "encodeDecode_out", 1, file_num);
receiver.Run();
receiver.Teardown();
rtpFile.Close();
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 8acacb9..8a20998 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -16,7 +16,8 @@
#include "absl/strings/string_view.h"
#include "api/environment/environment.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/RTPFile.h"
@@ -76,7 +77,7 @@
public:
Receiver();
virtual ~Receiver() {}
- void Setup(acm2::AcmReceiver* acm_receiver,
+ void Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
@@ -94,7 +95,8 @@
bool _firstTime;
protected:
- acm2::AcmReceiver* _acm_receiver;
+ NetEq* _neteq;
+ acm2::ResamplerHelper _resampler_helper;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
RTPHeader _rtpHeader;
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 9226b0f..24bca73 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -17,6 +17,7 @@
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/units/timestamp.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -30,7 +31,7 @@
lost_packet_counter_(0),
burst_lost_counter_(burst_length_) {}
-void ReceiverWithPacketLoss::Setup(acm2::AcmReceiver* acm_receiver,
+void ReceiverWithPacketLoss::Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
@@ -42,7 +43,7 @@
burst_lost_counter_ = burst_length_; // To prevent first packet gets lost.
rtc::StringBuilder ss;
ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
- Receiver::Setup(acm_receiver, rtpStream, ss.str(), channels, file_num);
+ Receiver::Setup(neteq, rtpStream, ss.str(), channels, file_num);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
@@ -61,10 +62,10 @@
}
if (!PacketLost()) {
- _acm_receiver->InsertPacket(_rtpHeader,
- rtc::ArrayView<const uint8_t>(
- _incomingPayload, _realPayloadSizeBytes),
- Timestamp::Millis(_nextTime));
+ _neteq->InsertPacket(_rtpHeader,
+ rtc::ArrayView<const uint8_t>(_incomingPayload,
+ _realPayloadSizeBytes),
+ Timestamp::Millis(_nextTime));
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
@@ -162,11 +163,10 @@
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
- std::unique_ptr<acm2::AcmReceiver> acm_receiver =
- std::make_unique<acm2::AcmReceiver>(
- env, acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()));
+ std::unique_ptr<NetEq> neteq = DefaultNetEqFactory().Create(
+ env, NetEq::Config(), CreateBuiltinAudioDecoderFactory());
ReceiverWithPacketLoss receiver;
- receiver.Setup(acm_receiver.get(), &rtpFile, "packetLoss_out", channels_, 15,
+ receiver.Setup(neteq.get(), &rtpFile, "packetLoss_out", channels_, 15,
actual_loss_rate_, burst_length_);
receiver.Run();
receiver.Teardown();
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index b256393..41a62d5 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -21,7 +21,7 @@
class ReceiverWithPacketLoss : public Receiver {
public:
ReceiverWithPacketLoss();
- void Setup(acm2::AcmReceiver* acm_receiver,
+ void Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 41c4a3a..1a9887b 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -18,7 +18,10 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/logging.h"
#include "rtc_base/string_encode.h"
@@ -47,7 +50,7 @@
// Class for simulating packet handling.
TestPack::TestPack()
- : receiver_acm_(NULL),
+ : neteq_(NULL),
sequence_number_(0),
timestamp_diff_(0),
last_in_timestamp_(0),
@@ -56,9 +59,8 @@
TestPack::~TestPack() {}
-void TestPack::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
- receiver_acm_ = acm_receiver;
- return;
+void TestPack::RegisterReceiverNetEq(NetEq* neteq) {
+ neteq_ = neteq;
}
int32_t TestPack::SendData(AudioFrameType frame_type,
@@ -84,7 +86,7 @@
// Only run mono for all test cases.
memcpy(payload_data_, payload_data, payload_size);
- status = receiver_acm_->InsertPacket(
+ status = neteq_->InsertPacket(
rtp_header, rtc::ArrayView<const uint8_t>(payload_data_, payload_size),
/*receive_time=*/Timestamp::MinusInfinity());
@@ -110,9 +112,9 @@
TestAllCodecs::TestAllCodecs()
: env_(CreateEnvironment()),
acm_a_(AudioCodingModule::Create()),
- acm_b_(std::make_unique<acm2::AcmReceiver>(
- env_,
- acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
+ neteq_(DefaultNetEqFactory().Create(env_,
+ NetEq::Config(),
+ CreateBuiltinAudioDecoderFactory())),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
@@ -130,7 +132,7 @@
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
- acm_b_->SetCodecs({{107, {"L16", 8000, 1}},
+ neteq_->SetCodecs({{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{111, {"L16", 8000, 2}},
@@ -151,7 +153,7 @@
// Create and connect the channel
channel_a_to_b_ = new TestPack;
acm_a_->RegisterTransportCallback(channel_a_to_b_);
- channel_a_to_b_->RegisterReceiverACM(acm_b_.get());
+ channel_a_to_b_->RegisterReceiverNetEq(neteq_.get());
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
@@ -330,6 +332,7 @@
void TestAllCodecs::Run(TestPack* channel) {
AudioFrame audio_frame;
+ acm2::ResamplerHelper resampler_helper;
int32_t out_freq_hz = outfile_b_.SamplingFrequency();
size_t receive_size;
@@ -367,8 +370,9 @@
// Run received side of ACM.
bool muted;
- CHECK_ERROR(acm_b_->GetAudio(out_freq_hz, &audio_frame, &muted));
+ CHECK_ERROR(neteq_->GetAudio(&audio_frame, &muted));
ASSERT_FALSE(muted);
+ EXPECT_TRUE(resampler_helper.MaybeResample(out_freq_hz, &audio_frame));
// Write output speech to file.
outfile_b_.Write10MsData(audio_frame.data(),
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index e078f49..0712332 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -14,18 +14,19 @@
#include <memory>
#include "api/environment/environment.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
+class NetEq;
+
class TestPack : public AudioPacketizationCallback {
public:
TestPack();
~TestPack();
- void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
+ void RegisterReceiverNetEq(NetEq* neteq);
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
@@ -39,7 +40,7 @@
void reset_payload_size();
private:
- acm2::AcmReceiver* receiver_acm_;
+ NetEq* neteq_;
uint16_t sequence_number_;
uint8_t payload_data_[60 * 32 * 2 * 2];
uint32_t timestamp_diff_;
@@ -71,7 +72,7 @@
const Environment env_;
std::unique_ptr<AudioCodingModule> acm_a_;
- std::unique_ptr<acm2::AcmReceiver> acm_b_;
+ std::unique_ptr<NetEq> neteq_;
TestPack* channel_a_to_b_;
PCMFile infile_a_;
PCMFile outfile_b_;
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index b4d138b..c1d28e8 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -28,6 +28,7 @@
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -45,9 +46,9 @@
AudioDecoderL16,
AudioDecoderOpus>()),
_acmA(AudioCodingModule::Create()),
- _acm_receiver(std::make_unique<acm2::AcmReceiver>(
- env_,
- acm2::AcmReceiver::Config(decoder_factory_))),
+ _neteq(DefaultNetEqFactory().Create(env_,
+ NetEq::Config(),
+ decoder_factory_)),
_channelA2B(NULL),
_testCntr(0) {}
@@ -66,7 +67,7 @@
// Create and connect the channel
_channelA2B = new Channel;
_acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acm_receiver.get());
+ _channelA2B->RegisterReceiverNetEq(_neteq.get());
RegisterSendCodec(_acmA, {"L16", 8000, 1}, Vad::kVadAggressive, true);
@@ -166,7 +167,7 @@
}
}
acm->SetEncoder(std::move(encoder));
- _acm_receiver->SetCodecs(receive_codecs);
+ _neteq->SetCodecs(receive_codecs);
}
void TestRedFec::Run() {
@@ -181,7 +182,8 @@
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
bool muted;
- EXPECT_EQ(0, _acm_receiver->GetAudio(outFreqHzB, &audioFrame, &muted));
+ EXPECT_EQ(NetEq::kOK, _neteq->GetAudio(&audioFrame, &muted));
+ EXPECT_TRUE(_resampler_helper.MaybeResample(outFreqHzB, &audioFrame));
ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_);
}
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index 56a6495..6235854 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -17,8 +17,9 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/environment/environment.h"
+#include "api/neteq/neteq.h"
#include "common_audio/vad/include/vad.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "test/scoped_key_value_config.h"
@@ -45,7 +46,8 @@
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<AudioCodingModule> _acmA;
- std::unique_ptr<acm2::AcmReceiver> _acm_receiver;
+ std::unique_ptr<NetEq> _neteq;
+ acm2::ResamplerHelper _resampler_helper;
Channel* _channelA2B;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 0c258e9..4561d4c 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -17,6 +17,7 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
@@ -26,7 +27,7 @@
// Class for simulating packet handling
TestPackStereo::TestPackStereo()
- : receiver_acm_(NULL),
+ : neteq_(NULL),
seq_no_(0),
timestamp_diff_(0),
last_in_timestamp_(0),
@@ -36,8 +37,8 @@
TestPackStereo::~TestPackStereo() {}
-void TestPackStereo::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
- receiver_acm_ = acm_receiver;
+void TestPackStereo::RegisterReceiverNetEq(NetEq* neteq) {
+ neteq_ = neteq;
return;
}
@@ -61,7 +62,7 @@
}
if (lost_packet_ == false) {
- status = receiver_acm_->InsertPacket(
+ status = neteq_->InsertPacket(
rtp_header, rtc::ArrayView<const uint8_t>(payload_data, payload_size),
/*receive_time=*/Timestamp::MinusInfinity());
@@ -101,9 +102,9 @@
TestStereo::TestStereo()
: env_(CreateEnvironment()),
acm_a_(AudioCodingModule::Create()),
- acm_b_(std::make_unique<acm2::AcmReceiver>(
- env_,
- acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
+ neteq_(DefaultNetEqFactory().Create(env_,
+ NetEq::Config(),
+ CreateBuiltinAudioDecoderFactory())),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),
@@ -136,10 +137,10 @@
in_file_mono_->ReadStereo(false);
// Create and initialize two ACMs, one for each side of a one-to-one call.
- ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
- acm_b_->FlushBuffers();
+ ASSERT_TRUE((acm_a_.get() != NULL) && (neteq_.get() != NULL));
+ neteq_->FlushBuffers();
- acm_b_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ neteq_->SetCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
@@ -162,7 +163,7 @@
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
EXPECT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_));
- channel_a2b_->RegisterReceiverACM(acm_b_.get());
+ channel_a2b_->RegisterReceiverNetEq(neteq_.get());
char codec_pcma_temp[] = "PCMA";
RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2);
@@ -400,7 +401,7 @@
OpenOutFile(test_cntr_);
// Encode and decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
- acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
+ neteq_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in stereo, decode in mono.
@@ -419,13 +420,13 @@
// Decode in stereo.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ neteq_->SetCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
Run(channel_a2b_, audio_channels, 2);
out_file_.Close();
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
+ neteq_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
@@ -468,7 +469,7 @@
case 'B': {
// We no longer use this case. Refactor code to avoid the switch.
ASSERT_TRUE(false);
- // my_acm = acm_b_.get();
+ // my_acm = neteq_.get();
break;
}
default:
@@ -571,7 +572,8 @@
// Run receive side of ACM
bool muted;
- EXPECT_EQ(0, acm_b_->GetAudio(out_freq_hz_b, &audio_frame, &muted));
+ EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&audio_frame, &muted));
+ EXPECT_TRUE(resampler_helper_.MaybeResample(out_freq_hz_b, &audio_frame));
ASSERT_FALSE(muted);
// Write output speech to file
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index b53187b..7f31569 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -16,7 +16,8 @@
#include <memory>
#include "api/environment/environment.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -31,7 +32,7 @@
TestPackStereo();
~TestPackStereo();
- void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
+ void RegisterReceiverNetEq(NetEq* neteq);
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
@@ -47,7 +48,7 @@
void set_lost_packet(bool lost);
private:
- acm2::AcmReceiver* receiver_acm_;
+ NetEq* neteq_;
int16_t seq_no_;
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
@@ -84,7 +85,8 @@
const Environment env_;
std::unique_ptr<AudioCodingModule> acm_a_;
- std::unique_ptr<acm2::AcmReceiver> acm_b_;
+ std::unique_ptr<NetEq> neteq_;
+ acm2::ResamplerHelper resampler_helper_;
TestPackStereo* channel_a2b_;
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index ac87ae1..ce3247c 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -22,6 +22,7 @@
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
@@ -73,16 +74,16 @@
decoder_factory_(
CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()),
acm_send_(AudioCodingModule::Create()),
- acm_receive_(std::make_unique<acm2::AcmReceiver>(
- env_,
- acm2::AcmReceiver::Config(decoder_factory_))),
+ neteq_(DefaultNetEqFactory().Create(env_,
+ NetEq::Config(),
+ decoder_factory_)),
channel_(std::make_unique<Channel>()),
packetization_callback_(
std::make_unique<MonitoringAudioPacketizationCallback>(
channel_.get())) {
EXPECT_EQ(
0, acm_send_->RegisterTransportCallback(packetization_callback_.get()));
- channel_->RegisterReceiverACM(acm_receive_.get());
+ channel_->RegisterReceiverNetEq(neteq_.get());
}
bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
@@ -106,7 +107,7 @@
acm_send_->SetEncoder(std::move(encoder));
std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
- acm_receive_->SetCodecs(receive_codecs);
+ neteq_->SetCodecs(receive_codecs);
return added_comfort_noise;
}
@@ -145,7 +146,8 @@
time_stamp_ += frame_size_samples;
EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
bool muted;
- acm_receive_->GetAudio(kOutputFreqHz, &audio_frame, &muted);
+ neteq_->GetAudio(&audio_frame, &muted);
+ resampler_helper_.MaybeResample(kOutputFreqHz, &audio_frame);
ASSERT_FALSE(muted);
out_file.Write10MsData(audio_frame);
}
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index b7f90a3..9d4f0af 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -17,8 +17,9 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/environment/environment.h"
+#include "api/neteq/neteq.h"
#include "common_audio/vad/include/vad.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/test/Channel.h"
@@ -87,7 +88,8 @@
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<AudioCodingModule> acm_send_;
- std::unique_ptr<acm2::AcmReceiver> acm_receive_;
+ std::unique_ptr<NetEq> neteq_;
+ acm2::ResamplerHelper resampler_helper_;
std::unique_ptr<Channel> channel_;
std::unique_ptr<MonitoringAudioPacketizationCallback> packetization_callback_;
uint32_t time_stamp_ = 0x12345678;
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 95cfef5..6377304 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -16,6 +16,7 @@
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/test/TestStereo.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -23,9 +24,9 @@
namespace webrtc {
OpusTest::OpusTest()
- : acm_receiver_(std::make_unique<acm2::AcmReceiver>(
- CreateEnvironment(),
- acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
+ : neteq_(DefaultNetEqFactory().Create(CreateEnvironment(),
+ NetEq::Config(),
+ CreateBuiltinAudioDecoderFactory())),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
@@ -84,18 +85,18 @@
WebRtcOpus_DecoderInit(opus_mono_decoder_);
WebRtcOpus_DecoderInit(opus_stereo_decoder_);
- ASSERT_TRUE(acm_receiver_.get() != NULL);
- acm_receiver_->FlushBuffers();
+ ASSERT_TRUE(neteq_.get() != NULL);
+ neteq_->FlushBuffers();
// Register Opus stereo as receiving codec.
constexpr int kOpusPayloadType = 120;
const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
payload_type_ = kOpusPayloadType;
- acm_receiver_->SetCodecs({{kOpusPayloadType, kOpusFormatStereo}});
+ neteq_->SetCodecs({{kOpusPayloadType, kOpusFormatStereo}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
- channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
+ channel_a2b_->RegisterReceiverNetEq(neteq_.get());
//
// Test Stereo.
@@ -156,7 +157,7 @@
// Register Opus mono as receiving codec.
const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
- acm_receiver_->SetCodecs({{kOpusPayloadType, kOpusFormatMono}});
+ neteq_->SetCodecs({{kOpusPayloadType, kOpusFormatMono}});
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 120);
@@ -355,8 +356,8 @@
// Run received side of ACM.
bool muted;
- ASSERT_EQ(0, acm_receiver_->GetAudio(out_freq_hz_b, &audio_frame, &muted));
- ASSERT_FALSE(muted);
+ ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&audio_frame, &muted));
+ ASSERT_TRUE(resampler_helper_.MaybeResample(out_freq_hz_b, &audio_frame));
// Write output speech to file.
out_file_.Write10MsData(
diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h
index cf5581a..a2ae758 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -15,7 +15,7 @@
#include <memory>
-#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -39,7 +39,8 @@
void OpenOutFile(int test_number);
- std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
+ std::unique_ptr<NetEq> neteq_;
+ acm2::ResamplerHelper resampler_helper_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index cb15610..3d62aab 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -15,9 +15,9 @@
#include "api/environment/environment_factory.h"
#include "api/rtp_headers.h"
#include "api/units/timestamp.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -26,9 +26,10 @@
class TargetDelayTest : public ::testing::Test {
protected:
TargetDelayTest()
- : receiver_(
- CreateEnvironment(),
- acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory())) {}
+ : neteq_(
+ DefaultNetEqFactory().Create(CreateEnvironment(),
+ NetEq::Config(),
+ CreateBuiltinAudioDecoderFactory())) {}
~TargetDelayTest() {}
@@ -36,7 +37,7 @@
constexpr int pltype = 108;
std::map<int, SdpAudioFormat> receive_codecs = {
{pltype, {"L16", kSampleRateHz, 1}}};
- receiver_.SetCodecs(receive_codecs);
+ neteq_->SetCodecs(receive_codecs);
rtp_header_.payloadType = pltype;
rtp_header_.timestamp = 0;
@@ -52,20 +53,20 @@
}
void OutOfRangeInput() {
- EXPECT_EQ(-1, SetMinimumDelay(-1));
- EXPECT_EQ(-1, SetMinimumDelay(10001));
+ EXPECT_FALSE(SetMinimumDelay(-1));
+ EXPECT_FALSE(SetMinimumDelay(10001));
}
void TargetDelayBufferMinMax() {
const int kTargetMinDelayMs = kNum10msPerFrame * 10;
- ASSERT_EQ(0, SetMinimumDelay(kTargetMinDelayMs));
+ ASSERT_TRUE(SetMinimumDelay(kTargetMinDelayMs));
for (int m = 0; m < 30; ++m) // Run enough iterations to fill the buffer.
Run(true);
int clean_optimal_delay = GetCurrentOptimalDelayMs();
EXPECT_EQ(kTargetMinDelayMs, clean_optimal_delay);
const int kTargetMaxDelayMs = 2 * (kNum10msPerFrame * 10);
- ASSERT_EQ(0, SetMaximumDelay(kTargetMaxDelayMs));
+ ASSERT_TRUE(SetMaximumDelay(kTargetMaxDelayMs));
for (int n = 0; n < 30; ++n) // Run enough iterations to fill the buffer.
Run(false);
@@ -86,10 +87,10 @@
void Push() {
rtp_header_.timestamp += kFrameSizeSamples;
rtp_header_.sequenceNumber++;
- ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
- rtc::ArrayView<const uint8_t>(
- payload_, kFrameSizeSamples * 2),
- Timestamp::MinusInfinity()));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_header_,
+ rtc::ArrayView<const uint8_t>(
+ payload_, kFrameSizeSamples * 2),
+ Timestamp::MinusInfinity()));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@@ -97,7 +98,7 @@
AudioFrame frame;
bool muted;
for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame.
- ASSERT_EQ(0, receiver_.GetAudio(-1, &frame, &muted));
+ ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&frame, &muted));
ASSERT_FALSE(muted);
// Had to use ASSERT_TRUE, ASSERT_EQ generated error.
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
@@ -124,20 +125,20 @@
}
int SetMinimumDelay(int delay_ms) {
- return receiver_.SetMinimumDelay(delay_ms);
+ return neteq_->SetMinimumDelay(delay_ms);
}
int SetMaximumDelay(int delay_ms) {
- return receiver_.SetMaximumDelay(delay_ms);
+ return neteq_->SetMaximumDelay(delay_ms);
}
int GetCurrentOptimalDelayMs() {
- NetworkStatistics stats;
- receiver_.GetNetworkStatistics(&stats);
- return stats.preferredBufferSize;
+ NetEqNetworkStatistics neteq_stats;
+ neteq_->NetworkStatistics(&neteq_stats);
+ return neteq_stats.preferred_buffer_size_ms;
}
- acm2::AcmReceiver receiver_;
+ std::unique_ptr<NetEq> neteq_;
RTPHeader rtp_header_;
uint8_t payload_[kPayloadLenBytes];
};
