Dedicated speed test for NetEq3
This is the same test as was aleready implemented for NetEq3 in r4763.
BUG=1363
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2234004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4782 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/neteq/neteq.gypi b/modules/audio_coding/neteq/neteq.gypi
index c94f7f8..31297ff 100644
--- a/modules/audio_coding/neteq/neteq.gypi
+++ b/modules/audio_coding/neteq/neteq.gypi
@@ -158,6 +158,20 @@
},
{
+ 'target_name': 'neteq3_speed_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'NetEq',
+ 'PCM16B',
+ 'neteq_unittest_tools',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [
+ 'test/neteq_speed_test.cc',
+ ],
+ },
+
+ {
'target_name': 'NetEqTestTools',
# Collection of useful functions used in other tests
'type': 'static_library',
diff --git a/modules/audio_coding/neteq/test/neteq_speed_test.cc b/modules/audio_coding/neteq/test/neteq_speed_test.cc
new file mode 100644
index 0000000..d1eaa0a
--- /dev/null
+++ b/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <iostream>
+
+#include "gflags/gflags.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
+#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
+#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+using webrtc::test::AudioLoop;
+using webrtc::test::RtpGenerator;
+using webrtc::WebRtcRTPHeader;
+
+// Flag validators.
+static bool ValidateRuntime(const char* flagname, int value) {
+ if (value > 0) // Value is ok.
+ return true;
+ printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+ return false;
+}
+static bool ValidateLossrate(const char* flagname, int value) {
+ if (value >= 0) // Value is ok.
+ return true;
+ printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+ return false;
+}
+static bool ValidateDriftfactor(const char* flagname, double value) {
+ if (value >= 0.0 && value < 1.0) // Value is ok.
+ return true;
+ printf("Invalid value for --%s: %f\n", flagname, value);
+ return false;
+}
+
+// Define command line flags.
+DEFINE_int32(runtime_ms, 10000, "Simulated runtime in ms.");
+static const bool runtime_ms_dummy =
+ google::RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
+DEFINE_int32(lossrate, 10,
+ "Packet lossrate; drop every N packets.");
+static const bool lossrate_dummy =
+ google::RegisterFlagValidator(&FLAGS_lossrate, &ValidateLossrate);
+DEFINE_double(drift, 0.1,
+ "Clockdrift factor.");
+static const bool drift_dummy =
+ google::RegisterFlagValidator(&FLAGS_drift, &ValidateDriftfactor);
+
+int main(int argc, char* argv[]) {
+ static const int kMaxChannels = 1;
+ static const int kMaxSamplesPerMs = 48000 / 1000;
+ static const int kOutputBlockSizeMs = 10;
+ const std::string kInputFileName =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ const int kSampRateHz = 32000;
+ const WebRtcNetEQDecoder kDecoderType = kDecoderPCM16Bswb32kHz;
+ const int kPayloadType = 95;
+
+ std::string program_name = argv[0];
+ std::string usage = "Tool for measuring the speed of NetEq.\n"
+ "Usage: " + program_name + " [options]\n\n"
+ " --runtime_ms=N runtime in ms; default is 10000 ms\n"
+ " --lossrate=N drop every N packets; default is 10\n"
+ " --drift=F clockdrift factor between 0.0 and 1.0; "
+ "default is 0.1\n";
+ google::SetUsageMessage(usage);
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ if (argc != 1) {
+ // Print usage information.
+ std::cout << google::ProgramUsage();
+ return 0;
+ }
+
+ // Initialize NetEq instance.
+ int error;
+ int inst_size_bytes;
+ error = WebRtcNetEQ_AssignSize(&inst_size_bytes);
+ if (error) {
+ std::cerr << "Error returned from WebRtcNetEQ_AssignSize." << std::endl;
+ exit(1);
+ }
+ char* inst_mem = new char[inst_size_bytes];
+ void* neteq_inst;
+ error = WebRtcNetEQ_Assign(&neteq_inst, inst_mem);
+ if (error) {
+ std::cerr << "Error returned from WebRtcNetEQ_Assign." << std::endl;
+ exit(1);
+ }
+ // Select decoders.
+ WebRtcNetEQDecoder decoder_list[] = {kDecoderType};
+ int max_number_of_packets;
+ int buffer_size_bytes;
+ int overhead_bytes_dummy;
+ error = WebRtcNetEQ_GetRecommendedBufferSize(
+ neteq_inst, decoder_list, sizeof(decoder_list) / sizeof(decoder_list[1]),
+ kTCPLargeJitter, &max_number_of_packets, &buffer_size_bytes,
+ &overhead_bytes_dummy);
+ if (error) {
+ std::cerr << "Error returned from WebRtcNetEQ_GetRecommendedBufferSize."
+ << std::endl;
+ exit(1);
+ }
+ char* buffer_mem = new char[buffer_size_bytes];
+ error = WebRtcNetEQ_AssignBuffer(neteq_inst, max_number_of_packets,
+ buffer_mem, buffer_size_bytes);
+ if (error) {
+ std::cerr << "Error returned from WebRtcNetEQ_AssignBuffer." << std::endl;
+ exit(1);
+ }
+ error = WebRtcNetEQ_Init(neteq_inst, kSampRateHz);
+ if (error) {
+ std::cerr << "Error returned from WebRtcNetEQ_Init." << std::endl;
+ exit(1);
+ }
+
+ // Register decoder.
+ WebRtcNetEQ_CodecDef codec_definition;
+ SET_CODEC_PAR(codec_definition, kDecoderType, kPayloadType, NULL,
+ kSampRateHz);
+ SET_PCM16B_SWB32_FUNCTIONS(codec_definition);
+ error = WebRtcNetEQ_CodecDbAdd(neteq_inst, &codec_definition);
+ if (error) {
+ std::cerr << "Cannot register decoder." << std::endl;
+ exit(1);
+ }
+
+ // Set up AudioLoop object.
+ AudioLoop audio_loop;
+ const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop.
+ const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms.
+ if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+ kInputBlockSizeSamples)) {
+ std::cerr << "Cannot initialize AudioLoop object." << std::endl;
+ exit(1);
+ }
+
+ int32_t time_now_ms = 0;
+
+ // Get first input packet.
+ WebRtcRTPHeader rtp_header;
+ RtpGenerator rtp_gen(kSampRateHz / 1000);
+ // Start with positive drift first half of simulation.
+ double drift_factor = 0.1;
+ rtp_gen.set_drift_factor(drift_factor);
+ bool drift_flipped = false;
+ int32_t packet_input_time_ms =
+ rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
+ const int16_t* input_samples = audio_loop.GetNextBlock();
+ if (!input_samples) exit(1);
+ uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
+ int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+ kInputBlockSizeSamples,
+ input_payload);
+ assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+
+ // Main loop.
+ while (time_now_ms < FLAGS_runtime_ms) {
+ while (packet_input_time_ms <= time_now_ms) {
+ // Drop every N packets, where N = FLAGS_lossrate.
+ bool lost = false;
+ if (FLAGS_lossrate > 0) {
+ lost = ((rtp_header.header.sequenceNumber - 1) % FLAGS_lossrate) == 0;
+ }
+ if (!lost) {
+ WebRtcNetEQ_RTPInfo rtp_info;
+ rtp_info.payloadType = rtp_header.header.payloadType;
+ rtp_info.sequenceNumber = rtp_header.header.sequenceNumber;
+ rtp_info.timeStamp = rtp_header.header.timestamp;
+ rtp_info.SSRC = rtp_header.header.ssrc;
+ rtp_info.markerBit = rtp_header.header.markerBit;
+ // Insert packet.
+ error = WebRtcNetEQ_RecInRTPStruct(
+ neteq_inst, &rtp_info, input_payload, payload_len,
+ packet_input_time_ms * kSampRateHz / 1000);
+ if (error != 0) {
+ std::cerr << "WebRtcNetEQ_RecInRTPStruct returned error code " <<
+ WebRtcNetEQ_GetErrorCode(neteq_inst) << std::endl;
+ exit(1);
+ }
+ }
+
+ // Get next packet.
+ packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
+ kInputBlockSizeSamples,
+ &rtp_header);
+ input_samples = audio_loop.GetNextBlock();
+ if (!input_samples) exit(1);
+ payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+ kInputBlockSizeSamples,
+ input_payload);
+ assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+ }
+
+ // Get output audio, but don't do anything with it.
+ static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
+ kMaxChannels;
+ int16_t out_data[kOutDataLen];
+ int16_t samples_per_channel;
+ error = WebRtcNetEQ_RecOut(neteq_inst, out_data, &samples_per_channel);
+ if (error != 0) {
+ std::cerr << "WebRtcNetEQ_RecOut returned error code " <<
+ WebRtcNetEQ_GetErrorCode(neteq_inst) << std::endl;
+ exit(1);
+ }
+ assert(samples_per_channel == kSampRateHz * 10 / 1000);
+
+ time_now_ms += kOutputBlockSizeMs;
+ if (time_now_ms >= FLAGS_runtime_ms / 2 && !drift_flipped) {
+ // Apply negative drift second half of simulation.
+ rtp_gen.set_drift_factor(-drift_factor);
+ drift_flipped = true;
+ }
+ }
+
+ std::cout << "Simulation done" << std::endl;
+ delete [] buffer_mem;
+ delete [] inst_mem;
+ return 0;
+}
diff --git a/modules/audio_coding/neteq4/test/neteq_speed_test.cc b/modules/audio_coding/neteq4/test/neteq_speed_test.cc
index d3fcb91..34f0de4 100644
--- a/modules/audio_coding/neteq4/test/neteq_speed_test.cc
+++ b/modules/audio_coding/neteq4/test/neteq_speed_test.cc
@@ -104,7 +104,6 @@
exit(1);
}
-
int32_t time_now_ms = 0;
// Get first input packet.
