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/*
* Copyright (c) 2017 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>
#ifdef WIN32
#include <winsock2.h>
#endif
#if defined(WEBRTC_LINUX) || defined(WEBRTC_FUCHSIA)
#include <netinet/in.h>
#endif
#include <iostream>
#include <map>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#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/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "rtc_base/numerics/safe_conversions.h"
ABSL_FLAG(bool, list_codecs, false, "Enumerate all codecs");
ABSL_FLAG(std::string, codec, "opus", "Codec to use");
ABSL_FLAG(int,
frame_len,
0,
"Frame length in ms; 0 indicates codec default value");
ABSL_FLAG(int, bitrate, 0, "Bitrate in bps; 0 indicates codec default value");
ABSL_FLAG(int,
payload_type,
-1,
"RTP payload type; -1 indicates codec default value");
ABSL_FLAG(int,
cng_payload_type,
-1,
"RTP payload type for CNG; -1 indicates default value");
ABSL_FLAG(int, ssrc, 0, "SSRC to write to the RTP header");
ABSL_FLAG(bool, dtx, false, "Use DTX/CNG");
ABSL_FLAG(int, sample_rate, 48000, "Sample rate of the input file");
ABSL_FLAG(bool, fec, false, "Use Opus FEC");
ABSL_FLAG(int, expected_loss, 0, "Expected packet loss percentage");
namespace webrtc {
namespace test {
namespace {
// Add new codecs here, and to the map below.
enum class CodecType {
kOpus,
kPcmU,
kPcmA,
kG722,
kPcm16b8,
kPcm16b16,
kPcm16b32,
kPcm16b48,
kIlbc,
};
struct CodecTypeAndInfo {
CodecType type;
int default_payload_type;
bool internal_dtx;
};
// List all supported codecs here. This map defines the command-line parameter
// value (the key string) for selecting each codec, together with information
// whether it is using internal or external DTX/CNG.
const std::map<std::string, CodecTypeAndInfo>& CodecList() {
static const auto* const codec_list =
new std::map<std::string, CodecTypeAndInfo>{
{"opus", {CodecType::kOpus, 111, true}},
{"pcmu", {CodecType::kPcmU, 0, false}},
{"pcma", {CodecType::kPcmA, 8, false}},
{"g722", {CodecType::kG722, 9, false}},
{"pcm16b_8", {CodecType::kPcm16b8, 93, false}},
{"pcm16b_16", {CodecType::kPcm16b16, 94, false}},
{"pcm16b_32", {CodecType::kPcm16b32, 95, false}},
{"pcm16b_48", {CodecType::kPcm16b48, 96, false}},
{"ilbc", {CodecType::kIlbc, 102, false}}};
return *codec_list;
}
// This class will receive callbacks from ACM when a packet is ready, and write
// it to the output file.
class Packetizer : public AudioPacketizationCallback {
public:
Packetizer(FILE* out_file, uint32_t ssrc, int timestamp_rate_hz)
: out_file_(out_file),
ssrc_(ssrc),
timestamp_rate_hz_(timestamp_rate_hz) {}
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
if (payload_len_bytes == 0) {
return 0;
}
constexpr size_t kRtpHeaderLength = 12;
constexpr size_t kRtpDumpHeaderLength = 8;
const uint16_t length = htons(rtc::checked_cast<uint16_t>(
kRtpHeaderLength + kRtpDumpHeaderLength + payload_len_bytes));
const uint16_t plen = htons(
rtc::checked_cast<uint16_t>(kRtpHeaderLength + payload_len_bytes));
const uint32_t offset = htonl(timestamp / (timestamp_rate_hz_ / 1000));
RTC_CHECK_EQ(fwrite(&length, sizeof(uint16_t), 1, out_file_), 1);
RTC_CHECK_EQ(fwrite(&plen, sizeof(uint16_t), 1, out_file_), 1);
RTC_CHECK_EQ(fwrite(&offset, sizeof(uint32_t), 1, out_file_), 1);
const uint8_t rtp_header[] = {0x80,
static_cast<uint8_t>(payload_type & 0x7F),
static_cast<uint8_t>(sequence_number_ >> 8),
static_cast<uint8_t>(sequence_number_),
static_cast<uint8_t>(timestamp >> 24),
static_cast<uint8_t>(timestamp >> 16),
static_cast<uint8_t>(timestamp >> 8),
static_cast<uint8_t>(timestamp),
static_cast<uint8_t>(ssrc_ >> 24),
static_cast<uint8_t>(ssrc_ >> 16),
static_cast<uint8_t>(ssrc_ >> 8),
static_cast<uint8_t>(ssrc_)};
static_assert(sizeof(rtp_header) == kRtpHeaderLength, "");
RTC_CHECK_EQ(
fwrite(rtp_header, sizeof(uint8_t), kRtpHeaderLength, out_file_),
kRtpHeaderLength);
++sequence_number_; // Intended to wrap on overflow.
RTC_CHECK_EQ(
fwrite(payload_data, sizeof(uint8_t), payload_len_bytes, out_file_),
payload_len_bytes);
return 0;
}
private:
FILE* const out_file_;
const uint32_t ssrc_;
const int timestamp_rate_hz_;
uint16_t sequence_number_ = 0;
};
void SetFrameLenIfFlagIsPositive(int* config_frame_len) {
if (absl::GetFlag(FLAGS_frame_len) > 0) {
*config_frame_len = absl::GetFlag(FLAGS_frame_len);
}
}
template <typename T>
typename T::Config GetCodecConfig() {
typename T::Config config;
SetFrameLenIfFlagIsPositive(&config.frame_size_ms);
RTC_CHECK(config.IsOk());
return config;
}
AudioEncoderL16::Config Pcm16bConfig(CodecType codec_type) {
auto config = GetCodecConfig<AudioEncoderL16>();
switch (codec_type) {
case CodecType::kPcm16b8:
config.sample_rate_hz = 8000;
return config;
case CodecType::kPcm16b16:
config.sample_rate_hz = 16000;
return config;
case CodecType::kPcm16b32:
config.sample_rate_hz = 32000;
return config;
case CodecType::kPcm16b48:
config.sample_rate_hz = 48000;
return config;
default:
RTC_DCHECK_NOTREACHED();
return config;
}
}
std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
int payload_type) {
switch (codec_type) {
case CodecType::kOpus: {
AudioEncoderOpus::Config config = GetCodecConfig<AudioEncoderOpus>();
if (absl::GetFlag(FLAGS_bitrate) > 0) {
config.bitrate_bps = absl::GetFlag(FLAGS_bitrate);
}
config.dtx_enabled = absl::GetFlag(FLAGS_dtx);
config.fec_enabled = absl::GetFlag(FLAGS_fec);
RTC_CHECK(config.IsOk());
return AudioEncoderOpus::MakeAudioEncoder(CreateEnvironment(), config,
{.payload_type = payload_type});
}
case CodecType::kPcmU:
case CodecType::kPcmA: {
AudioEncoderG711::Config config = GetCodecConfig<AudioEncoderG711>();
config.type = codec_type == CodecType::kPcmU
? AudioEncoderG711::Config::Type::kPcmU
: AudioEncoderG711::Config::Type::kPcmA;
RTC_CHECK(config.IsOk());
return AudioEncoderG711::MakeAudioEncoder(config, payload_type);
}
case CodecType::kG722: {
return AudioEncoderG722::MakeAudioEncoder(
GetCodecConfig<AudioEncoderG722>(), payload_type);
}
case CodecType::kPcm16b8:
case CodecType::kPcm16b16:
case CodecType::kPcm16b32:
case CodecType::kPcm16b48: {
return AudioEncoderL16::MakeAudioEncoder(Pcm16bConfig(codec_type),
payload_type);
}
case CodecType::kIlbc: {
return AudioEncoderIlbc::MakeAudioEncoder(
GetCodecConfig<AudioEncoderIlbc>(), payload_type);
}
}
RTC_DCHECK_NOTREACHED();
return nullptr;
}
AudioEncoderCngConfig GetCngConfig(int sample_rate_hz) {
AudioEncoderCngConfig cng_config;
const auto default_payload_type = [&] {
switch (sample_rate_hz) {
case 8000:
return 13;
case 16000:
return 98;
case 32000:
return 99;
case 48000:
return 100;
default:
RTC_DCHECK_NOTREACHED();
}
return 0;
};
cng_config.payload_type = absl::GetFlag(FLAGS_cng_payload_type) != -1
? absl::GetFlag(FLAGS_cng_payload_type)
: default_payload_type();
return cng_config;
}
int RunRtpEncode(int argc, char* argv[]) {
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
const std::string usage =
"Tool for generating an RTP dump file from audio input.\n"
"Example usage:\n"
"./rtp_encode input.pcm output.rtp --codec=[codec] "
"--frame_len=[frame_len] --bitrate=[bitrate]\n\n";
if (!absl::GetFlag(FLAGS_list_codecs) && args.size() != 3) {
printf("%s", usage.c_str());
return 1;
}
if (absl::GetFlag(FLAGS_list_codecs)) {
printf("The following arguments are valid --codec parameters:\n");
for (const auto& c : CodecList()) {
printf(" %s\n", c.first.c_str());
}
return 0;
}
const auto codec_it = CodecList().find(absl::GetFlag(FLAGS_codec));
if (codec_it == CodecList().end()) {
printf("%s is not a valid codec name.\n",
absl::GetFlag(FLAGS_codec).c_str());
printf("Use argument --list_codecs to see all valid codec names.\n");
return 1;
}
// Create the codec.
const int payload_type = absl::GetFlag(FLAGS_payload_type) == -1
? codec_it->second.default_payload_type
: absl::GetFlag(FLAGS_payload_type);
std::unique_ptr<AudioEncoder> codec =
CreateEncoder(codec_it->second.type, payload_type);
// Create an external VAD/CNG encoder if needed.
if (absl::GetFlag(FLAGS_dtx) && !codec_it->second.internal_dtx) {
AudioEncoderCngConfig cng_config = GetCngConfig(codec->SampleRateHz());
RTC_DCHECK(codec);
cng_config.speech_encoder = std::move(codec);
codec = CreateComfortNoiseEncoder(std::move(cng_config));
}
RTC_DCHECK(codec);
// Set up ACM.
const int timestamp_rate_hz = codec->RtpTimestampRateHz();
auto acm(AudioCodingModule::Create());
acm->SetEncoder(std::move(codec));
acm->SetPacketLossRate(absl::GetFlag(FLAGS_expected_loss));
// Open files.
printf("Input file: %s\n", args[1]);
InputAudioFile input_file(args[1], false); // Open input in non-looping mode.
FILE* out_file = fopen(args[2], "wb");
RTC_CHECK(out_file) << "Could not open file " << args[2] << " for writing";
printf("Output file: %s\n", args[2]);
fprintf(out_file, "#!rtpplay1.0 \n"); //,
// Write 3 32-bit values followed by 2 16-bit values, all set to 0. This means
// a total of 16 bytes.
const uint8_t file_header[16] = {0};
RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
// Create and register the packetizer, which will write the packets to file.
Packetizer packetizer(out_file, absl::GetFlag(FLAGS_ssrc), timestamp_rate_hz);
RTC_DCHECK_EQ(acm->RegisterTransportCallback(&packetizer), 0);
AudioFrame audio_frame;
audio_frame.samples_per_channel_ =
absl::GetFlag(FLAGS_sample_rate) / 100; // 10 ms
audio_frame.sample_rate_hz_ = absl::GetFlag(FLAGS_sample_rate);
audio_frame.num_channels_ = 1;
while (input_file.Read(audio_frame.samples_per_channel_,
audio_frame.mutable_data())) {
RTC_CHECK_GE(acm->Add10MsData(audio_frame), 0);
audio_frame.timestamp_ += audio_frame.samples_per_channel_;
}
return 0;
}
} // namespace
} // namespace test
} // namespace webrtc
int main(int argc, char* argv[]) {
return webrtc::test::RunRtpEncode(argc, argv);
}