Use test::Packet test::PacketSource classes in neteq_rtpplay
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.
Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.
Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).
BUG=2692
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/neteq/neteq_tests.gypi b/modules/audio_coding/neteq/neteq_tests.gypi
index 97d835f..d134dcd 100644
--- a/modules/audio_coding/neteq/neteq_tests.gypi
+++ b/modules/audio_coding/neteq/neteq_tests.gypi
@@ -13,7 +13,6 @@
'type': 'executable',
'dependencies': [
'neteq',
- 'neteq_test_tools',
'neteq_unittest_tools',
'PCM16B',
'<(webrtc_root)/test/test.gyp:test_support_main',
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 3e3540a..e144ba7 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -21,9 +21,9 @@
#include "google/gflags.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -92,8 +92,6 @@
google::RegisterFlagValidator(&FLAGS_cn_swb48, &ValidatePayloadType);
DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
"codec");
-DEFINE_bool(dummy_rtp, false, "The input file contains ""dummy"" RTP data, "
- "i.e., only headers");
DEFINE_string(replacement_audio_file, "",
"A PCM file that will be used to populate ""dummy"" RTP packets");
@@ -107,7 +105,7 @@
size_t* payload_mem_size_bytes,
size_t* frame_size_samples,
WebRtcRTPHeader* rtp_header,
- NETEQTEST_RTPpacket* next_rtp);
+ const webrtc::test::Packet* next_packet);
int CodecSampleRate(uint8_t payload_type);
int CodecTimestampRate(uint8_t payload_type);
bool IsComfortNosie(uint8_t payload_type);
@@ -139,15 +137,13 @@
return 0;
}
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- std::cerr << "Cannot open input file " << argv[1] << std::endl;
- exit(1);
- }
- std::cout << "Input file: " << argv[1] << std::endl;
+ printf("Input file: %s\n", argv[1]);
+ webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+ webrtc::test::RtpFileSource::Create(argv[1]));
+ assert(file_source.get());
FILE* out_file = fopen(argv[2], "wb");
- if (!in_file) {
+ if (!out_file) {
std::cerr << "Cannot open output file " << argv[2] << std::endl;
exit(1);
}
@@ -162,12 +158,6 @@
replace_payload = true;
}
- // Read RTP file header.
- if (NETEQTEST_RTPpacket::skipFileHeader(in_file) != 0) {
- std::cerr << "Wrong format in RTP file" << std::endl;
- exit(1);
- }
-
// Enable tracing.
webrtc::Trace::CreateTrace();
webrtc::Trace::SetTraceFile((webrtc::test::OutputPath() +
@@ -182,25 +172,17 @@
RegisterPayloadTypes(neteq);
// Read first packet.
- NETEQTEST_RTPpacket* rtp;
- NETEQTEST_RTPpacket* next_rtp = NULL;
- if (!FLAGS_dummy_rtp) {
- rtp = new NETEQTEST_RTPpacket();
- if (replace_payload) {
- next_rtp = new NETEQTEST_RTPpacket();
- }
- } else {
- rtp = new NETEQTEST_DummyRTPpacket();
- if (replace_payload) {
- next_rtp = new NETEQTEST_DummyRTPpacket();
- }
+ if (file_source->EndOfFile()) {
+ printf("Warning: RTP file is empty");
+ webrtc::Trace::ReturnTrace();
+ return 0;
}
- rtp->readFromFile(in_file);
- if (rtp->dataLen() < 0) {
- std::cout << "Warning: RTP file is empty" << std::endl;
- }
+ webrtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
+ bool packet_available = true;
// Set up variables for audio replacement if needed.
+ webrtc::scoped_ptr<webrtc::test::Packet> next_packet;
+ bool next_packet_available = false;
size_t input_frame_size_timestamps = 0;
webrtc::scoped_ptr<int16_t[]> replacement_audio;
webrtc::scoped_ptr<uint8_t[]> payload;
@@ -213,13 +195,15 @@
replacement_audio.reset(new int16_t[input_frame_size_timestamps]);
payload_mem_size_bytes = 2 * input_frame_size_timestamps;
payload.reset(new uint8_t[payload_mem_size_bytes]);
- assert(next_rtp);
- next_rtp->readFromFile(in_file);
+ assert(!file_source->EndOfFile());
+ next_packet.reset(file_source->NextPacket());
+ next_packet_available = true;
}
// This is the main simulation loop.
- int time_now_ms = rtp->time(); // Start immediately with the first packet.
- int next_input_time_ms = rtp->time();
+ // Set the simulation clock to start immediately with the first packet.
+ int time_now_ms = packet->time_ms();
+ int next_input_time_ms = time_now_ms;
int next_output_time_ms = time_now_ms;
if (time_now_ms % kOutputBlockSizeMs != 0) {
// Make sure that next_output_time_ms is rounded up to the next multiple
@@ -227,43 +211,52 @@
next_output_time_ms +=
kOutputBlockSizeMs - time_now_ms % kOutputBlockSizeMs;
}
- while (rtp->dataLen() >= 0) {
+ while (packet_available) {
// Check if it is time to insert packet.
- while (time_now_ms >= next_input_time_ms && rtp->dataLen() >= 0) {
- if (rtp->dataLen() > 0) {
- // Parse RTP header.
- WebRtcRTPHeader rtp_header;
- rtp->parseHeader(&rtp_header);
- uint8_t* payload_ptr = rtp->payload();
- size_t payload_len = rtp->payloadLen();
- if (replace_payload) {
- payload_len = ReplacePayload(replacement_audio_file.get(),
- &replacement_audio,
- &payload,
- &payload_mem_size_bytes,
- &input_frame_size_timestamps,
- &rtp_header,
- next_rtp);
- payload_ptr = payload.get();
- }
- int error = neteq->InsertPacket(rtp_header, payload_ptr,
- static_cast<int>(payload_len),
- rtp->time() * sample_rate_hz / 1000);
- if (error != NetEq::kOK) {
- std::cerr << "InsertPacket returned error code " <<
- neteq->LastError() << std::endl;
- }
- }
- // Get next packet from file.
- rtp->readFromFile(in_file);
+ while (time_now_ms >= next_input_time_ms && packet_available) {
+ assert(packet->virtual_payload_length_bytes() > 0);
+ // Parse RTP header.
+ WebRtcRTPHeader rtp_header;
+ packet->ConvertHeader(&rtp_header);
+ const uint8_t* payload_ptr = packet->payload();
+ size_t payload_len = packet->payload_length_bytes();
if (replace_payload) {
- // At this point |rtp| contains the packet *after* |next_rtp|.
- // Swap RTP packet objects between |rtp| and |next_rtp|.
- NETEQTEST_RTPpacket* temp_rtp = rtp;
- rtp = next_rtp;
- next_rtp = temp_rtp;
+ payload_len = ReplacePayload(replacement_audio_file.get(),
+ &replacement_audio,
+ &payload,
+ &payload_mem_size_bytes,
+ &input_frame_size_timestamps,
+ &rtp_header,
+ next_packet.get());
+ payload_ptr = payload.get();
}
- next_input_time_ms = rtp->time();
+ int error =
+ neteq->InsertPacket(rtp_header,
+ payload_ptr,
+ static_cast<int>(payload_len),
+ packet->time_ms() * sample_rate_hz / 1000);
+ if (error != NetEq::kOK) {
+ std::cerr << "InsertPacket returned error code " << neteq->LastError()
+ << std::endl;
+ }
+
+ // Get next packet from file.
+ if (!file_source->EndOfFile()) {
+ packet.reset(file_source->NextPacket());
+ } else {
+ packet_available = false;
+ }
+ if (replace_payload) {
+ // At this point |packet| contains the packet *after* |next_packet|.
+ // Swap Packet objects between |packet| and |next_packet|.
+ packet.swap(next_packet);
+ // Swap the status indicators unless they're already the same.
+ if (packet_available != next_packet_available) {
+ packet_available = !packet_available;
+ next_packet_available = !next_packet_available;
+ }
+ }
+ next_input_time_ms = packet->time_ms();
}
// Check if it is time to get output audio.
@@ -300,10 +293,7 @@
std::cout << "Simulation done" << std::endl;
- fclose(in_file);
fclose(out_file);
- delete rtp;
- delete next_rtp;
delete neteq;
webrtc::Trace::ReturnTrace();
return 0;
@@ -503,7 +493,7 @@
size_t* payload_mem_size_bytes,
size_t* frame_size_samples,
WebRtcRTPHeader* rtp_header,
- NETEQTEST_RTPpacket* next_rtp) {
+ const webrtc::test::Packet* next_packet) {
size_t payload_len = 0;
// Check for CNG.
if (IsComfortNosie(rtp_header->header.payloadType)) {
@@ -515,18 +505,18 @@
(*payload)[0] = 127; // Max attenuation of CNG.
payload_len = 1;
} else {
- if (next_rtp->payloadLen() > 0) {
- // Check if payload length has changed.
- if (next_rtp->sequenceNumber() == rtp_header->header.sequenceNumber + 1) {
- if (*frame_size_samples !=
- next_rtp->timeStamp() - rtp_header->header.timestamp) {
- *frame_size_samples =
- next_rtp->timeStamp() - rtp_header->header.timestamp;
- (*replacement_audio).reset(
- new int16_t[*frame_size_samples]);
- *payload_mem_size_bytes = 2 * *frame_size_samples;
- (*payload).reset(new uint8_t[*payload_mem_size_bytes]);
- }
+ assert(next_packet->virtual_payload_length_bytes() > 0);
+ // Check if payload length has changed.
+ if (next_packet->header().sequenceNumber ==
+ rtp_header->header.sequenceNumber + 1) {
+ if (*frame_size_samples !=
+ next_packet->header().timestamp - rtp_header->header.timestamp) {
+ *frame_size_samples =
+ next_packet->header().timestamp - rtp_header->header.timestamp;
+ (*replacement_audio).reset(
+ new int16_t[*frame_size_samples]);
+ *payload_mem_size_bytes = 2 * *frame_size_samples;
+ (*payload).reset(new uint8_t[*payload_mem_size_bytes]);
}
}
// Get new speech.
@@ -545,7 +535,7 @@
assert(*frame_size_samples > 0);
if (!replacement_audio_file->Read(*frame_size_samples,
(*replacement_audio).get())) {
- std::cerr << "Could no read replacement audio file." << std::endl;
+ std::cerr << "Could not read replacement audio file." << std::endl;
webrtc::Trace::ReturnTrace();
exit(1);
}
diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc
index d8fb713..794c308 100644
--- a/modules/audio_coding/neteq/tools/packet.cc
+++ b/modules/audio_coding/neteq/tools/packet.cc
@@ -9,6 +9,10 @@
*/
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+
+#include <string.h>
+
+#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
namespace webrtc {
@@ -117,6 +121,14 @@
}
}
+void Packet::ConvertHeader(WebRtcRTPHeader* copy_to) const {
+ memcpy(©_to->header, &header_, sizeof(header_));
+ copy_to->frameType = kAudioFrameSpeech;
+ copy_to->type.Audio.numEnergy = 0;
+ copy_to->type.Audio.channel = 1;
+ copy_to->type.Audio.isCNG = false;
+}
+
bool Packet::ParseHeader(const RtpHeaderParser& parser) {
bool valid_header = parser.Parse(
payload_memory_.get(), static_cast<int>(packet_length_bytes_), &header_);
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index eb8ce28..df7aeb7 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -21,6 +21,7 @@
namespace webrtc {
class RtpHeaderParser;
+struct WebRtcRTPHeader;
namespace test {
@@ -89,6 +90,10 @@
const RTPHeader& header() const { return header_; }
+ // Copies the packet header information, converting from the native RTPHeader
+ // type to WebRtcRTPHeader.
+ void ConvertHeader(WebRtcRTPHeader* copy_to) const;
+
void set_time_ms(double time) { time_ms_ = time; }
double time_ms() const { return time_ms_; }
bool valid_header() const { return valid_header_; }
diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc
index 773cc2c..b07de0b 100644
--- a/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -59,11 +59,6 @@
return 0;
}
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[1]);
- return -1;
- }
printf("Input file: %s\n", argv[1]);
webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
@@ -140,7 +135,6 @@
}
}
- fclose(in_file);
fclose(out_file);
return 0;
