diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc index f0dc2cb6f9..b3f6a775a6 100644 --- a/webrtc/modules/audio_processing/test/process_test.cc +++ b/webrtc/modules/audio_processing/test/process_test.cc @@ -59,7 +59,7 @@ void usage() { "when -ir or -i is used, the specified files will be processed directly in\n" "a simulation mode. Otherwise the full set of legacy test files is expected\n" "to be present in the working directory. OUT_FILE should be specified\n" - "without extension to support both int and float output.\n\n"); + "without extension to support both raw and wav output.\n\n"); printf("Options\n"); printf("General configuration (only used for the simulation mode):\n"); printf(" -fs SAMPLE_RATE_HZ\n"); @@ -112,6 +112,7 @@ void usage() { printf(" --perf Measure performance.\n"); printf(" --quiet Suppress text output.\n"); printf(" --no_progress Suppress progress.\n"); + printf(" --raw_output Raw output instead of WAV file.\n"); printf(" --debug_file FILE Dump a debug recording.\n"); } @@ -167,6 +168,7 @@ void void_main(int argc, char* argv[]) { bool perf_testing = false; bool verbose = true; bool progress = true; + bool raw_output = false; int extra_delay_ms = 0; int override_delay_ms = 0; @@ -427,6 +429,9 @@ void void_main(int argc, char* argv[]) { } else if (strcmp(argv[i], "--no_progress") == 0) { progress = false; + } else if (strcmp(argv[i], "--raw_output") == 0) { + raw_output = true; + } else if (strcmp(argv[i], "--debug_file") == 0) { i++; ASSERT_LT(i, argc) << "Specify filename after --debug_file"; @@ -464,8 +469,6 @@ void void_main(int argc, char* argv[]) { if (out_filename.size() == 0) { out_filename = out_path + "out"; } - std::string out_float_filename = out_filename + ".float"; - out_filename += ".pcm"; if (!vad_out_filename) { vad_out_filename = vad_file_default.c_str(); @@ -486,6 +489,9 @@ void void_main(int argc, char* argv[]) { FILE* aecm_echo_path_in_file = NULL; FILE* aecm_echo_path_out_file = NULL; + scoped_ptr output_wav_file; + scoped_ptr output_raw_file; + if (pb_filename) { pb_file = OpenFile(pb_filename, "rb"); } else { @@ -628,6 +634,14 @@ void void_main(int argc, char* argv[]) { printf(" Reverse channels: %d\n", msg.num_reverse_channels()); } + if (!raw_output) { + // The WAV file needs to be reset every time, because it cant change + // it's sample rate or number of channels. + output_wav_file.reset(new WavFile(out_filename + ".wav", + output_sample_rate, + msg.num_output_channels())); + } + } else if (event_msg.type() == Event::REVERSE_STREAM) { ASSERT_TRUE(event_msg.has_reverse_stream()); ReverseStream msg = event_msg.reverse_stream(); @@ -772,20 +786,24 @@ void void_main(int argc, char* argv[]) { } } - size_t num_samples = - apm->num_output_channels() * output_sample_rate / 100; + const size_t samples_per_channel = output_sample_rate / 100; if (msg.has_input_data()) { - static FILE* out_file = OpenFile(out_filename, "wb"); - ASSERT_EQ(num_samples, fwrite(near_frame.data_, - sizeof(*near_frame.data_), - num_samples, - out_file)); + if (raw_output && !output_raw_file) { + output_raw_file.reset(new RawFile(out_filename + ".pcm")); + } + WriteIntData(near_frame.data_, + apm->num_output_channels() * samples_per_channel, + output_wav_file.get(), + output_raw_file.get()); } else { - static FILE* out_float_file = OpenFile(out_float_filename, "wb"); - ASSERT_EQ(num_samples, fwrite(primary_cb->data(), - sizeof(*primary_cb->data()), - num_samples, - out_float_file)); + if (raw_output && !output_raw_file) { + output_raw_file.reset(new RawFile(out_filename + ".float")); + } + WriteFloatData(primary_cb->channels(), + samples_per_channel, + apm->num_output_channels(), + output_wav_file.get(), + output_raw_file.get()); } } } @@ -855,6 +873,14 @@ void void_main(int argc, char* argv[]) { near_frame.sample_rate_hz_ = sample_rate_hz; near_frame.samples_per_channel_ = samples_per_channel; + if (!raw_output) { + // The WAV file needs to be reset every time, because it cant change + // it's sample rate or number of channels. + output_wav_file.reset(new WavFile(out_filename + ".wav", + sample_rate_hz, + num_capture_output_channels)); + } + if (verbose) { printf("Init at frame: %d (primary), %d (reverse)\n", primary_count, reverse_count); @@ -999,12 +1025,13 @@ void void_main(int argc, char* argv[]) { } } - size = samples_per_channel * near_frame.num_channels_; - static FILE* out_file = OpenFile(out_filename, "wb"); - ASSERT_EQ(size, fwrite(near_frame.data_, - sizeof(int16_t), - size, - out_file)); + if (raw_output && !output_raw_file) { + output_raw_file.reset(new RawFile(out_filename + ".pcm")); + } + WriteIntData(near_frame.data_, + size, + output_wav_file.get(), + output_raw_file.get()); } else { FAIL() << "Event " << event << " is unrecognized"; diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h index e5204da5a0..61edd8f35b 100644 --- a/webrtc/modules/audio_processing/test/test_utils.h +++ b/webrtc/modules/audio_processing/test/test_utils.h @@ -8,7 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #include "webrtc/audio_processing/debug.pb.h" +#include "webrtc/common_audio/include/audio_util.h" +#include "webrtc/common_audio/wav_writer.h" #include "webrtc/modules/audio_processing/common.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/interface/module_common_types.h" @@ -19,6 +23,64 @@ namespace webrtc { static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError; #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr)) +class RawFile { + public: + RawFile(const std::string& filename) + : file_handle_(fopen(filename.c_str(), "wb")) {} + + ~RawFile() { + fclose(file_handle_); + } + + void WriteSamples(const int16_t* samples, size_t num_samples) { +#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +#error "Need to convert samples to little-endian when writing to PCM file" +#endif + fwrite(samples, sizeof(*samples), num_samples, file_handle_); + } + + void WriteSamples(const float* samples, size_t num_samples) { + fwrite(samples, sizeof(*samples), num_samples, file_handle_); + } + + private: + FILE* file_handle_; +}; + +static inline void WriteIntData(const int16_t* data, + size_t length, + WavFile* wav_file, + RawFile* raw_file) { + if (wav_file) { + wav_file->WriteSamples(data, length); + } + if (raw_file) { + raw_file->WriteSamples(data, length); + } +} + +static inline void WriteFloatData(const float* const* data, + size_t samples_per_channel, + int num_channels, + WavFile* wav_file, + RawFile* raw_file) { + size_t length = num_channels * samples_per_channel; + scoped_ptr buffer(new float[length]); + Interleave(data, samples_per_channel, num_channels, buffer.get()); + if (raw_file) { + raw_file->WriteSamples(buffer.get(), length); + } + // TODO(aluebs): Use ScaleToInt16Range() from audio_util + for (size_t i = 0; i < length; ++i) { + buffer[i] = buffer[i] > 0 ? + buffer[i] * std::numeric_limits::max() : + -buffer[i] * std::numeric_limits::min(); + } + if (wav_file) { + wav_file->WriteSamples(buffer.get(), length); + } +} + // Exits on failure; do not use in unit tests. static inline FILE* OpenFile(const std::string& filename, const char* mode) { FILE* file = fopen(filename.c_str(), mode); diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc index bb76a2dc60..249b6682f8 100644 --- a/webrtc/modules/audio_processing/test/unpack.cc +++ b/webrtc/modules/audio_processing/test/unpack.cc @@ -14,28 +14,19 @@ // to unpack the file into its component parts: audio and other data. #include -#include #include "gflags/gflags.h" #include "webrtc/audio_processing/debug.pb.h" -#include "webrtc/common_audio/include/audio_util.h" -#include "webrtc/common_audio/wav_writer.h" #include "webrtc/modules/audio_processing/test/test_utils.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" // TODO(andrew): unpack more of the data. -DEFINE_string(input_file, "input.pcm", "The name of the input stream file."); -DEFINE_string(input_wav_file, "input.wav", - "The name of the WAV input stream file."); -DEFINE_string(output_file, "ref_out.pcm", +DEFINE_string(input_file, "input", "The name of the input stream file."); +DEFINE_string(output_file, "ref_out", "The name of the reference output stream file."); -DEFINE_string(output_wav_file, "ref_out.wav", - "The name of the WAV reference output stream file."); -DEFINE_string(reverse_file, "reverse.pcm", +DEFINE_string(reverse_file, "reverse", "The name of the reverse input stream file."); -DEFINE_string(reverse_wav_file, "reverse.wav", - "The name of the WAV reverse input stream file."); DEFINE_string(delay_file, "delay.int32", "The name of the delay file."); DEFINE_string(drift_file, "drift.int32", "The name of the drift file."); DEFINE_string(level_file, "level.int32", "The name of the level file."); @@ -43,7 +34,7 @@ DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file."); DEFINE_string(settings_file, "settings.txt", "The name of the settings file."); DEFINE_bool(full, false, "Unpack the full set of files (normally not needed)."); -DEFINE_bool(pcm, false, "Write to PCM instead of WAV file."); +DEFINE_bool(raw, false, "Write raw data instead of a WAV file."); namespace webrtc { @@ -52,36 +43,6 @@ using audioproc::ReverseStream; using audioproc::Stream; using audioproc::Init; -class PcmFile { - public: - PcmFile(const std::string& filename) - : file_handle_(fopen(filename.c_str(), "wb")) {} - - ~PcmFile() { - fclose(file_handle_); - } - - void WriteSamples(const int16_t* samples, size_t num_samples) { -#ifndef WEBRTC_ARCH_LITTLE_ENDIAN -#error "Need to convert samples to little-endian when writing to PCM file" -#endif - fwrite(samples, sizeof(*samples), num_samples, file_handle_); - } - - void WriteSamples(const float* samples, size_t num_samples) { - static const size_t kChunksize = 4096 / sizeof(uint16_t); - for (size_t i = 0; i < num_samples; i += kChunksize) { - int16_t isamples[kChunksize]; - const size_t chunk = std::min(kChunksize, num_samples - i); - RoundToInt16(samples + i, chunk, isamples); - WriteSamples(isamples, chunk); - } - } - - private: - FILE* file_handle_; -}; - void WriteData(const void* data, size_t size, FILE* file, const std::string& filename) { if (fwrite(data, size, 1, file) != 1) { @@ -90,40 +51,6 @@ void WriteData(const void* data, size_t size, FILE* file, } } -void WriteIntData(const int16_t* data, - size_t length, - WavFile* wav_file, - PcmFile* pcm_file) { - if (wav_file) { - wav_file->WriteSamples(data, length); - } - if (pcm_file) { - pcm_file->WriteSamples(data, length); - } -} - -void WriteFloatData(const float* const* data, - size_t samples_per_channel, - int num_channels, - WavFile* wav_file, - PcmFile* pcm_file) { - size_t length = num_channels * samples_per_channel; - scoped_ptr buffer(new float[length]); - Interleave(data, samples_per_channel, num_channels, buffer.get()); - // TODO(aluebs): Use ScaleToInt16Range() from audio_util - for (size_t i = 0; i < length; ++i) { - buffer[i] = buffer[i] > 0 ? - buffer[i] * std::numeric_limits::max() : - -buffer[i] * std::numeric_limits::min(); - } - if (wav_file) { - wav_file->WriteSamples(buffer.get(), length); - } - if (pcm_file) { - pcm_file->WriteSamples(buffer.get(), length); - } -} - int do_main(int argc, char* argv[]) { std::string program_name = argv[0]; std::string usage = "Commandline tool to unpack audioproc debug files.\n" @@ -149,9 +76,9 @@ int do_main(int argc, char* argv[]) { scoped_ptr reverse_wav_file; scoped_ptr input_wav_file; scoped_ptr output_wav_file; - scoped_ptr reverse_pcm_file; - scoped_ptr input_pcm_file; - scoped_ptr output_pcm_file; + scoped_ptr reverse_raw_file; + scoped_ptr input_raw_file; + scoped_ptr output_raw_file; while (ReadMessageFromFile(debug_file, &event_msg)) { if (event_msg.type() == Event::REVERSE_STREAM) { if (!event_msg.has_reverse_stream()) { @@ -161,6 +88,9 @@ int do_main(int argc, char* argv[]) { const ReverseStream msg = event_msg.reverse_stream(); if (msg.has_data()) { + if (FLAGS_raw && !reverse_raw_file) { + reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm")); + } // TODO(aluebs): Replace "num_reverse_channels * // reverse_samples_per_channel" with "msg.data().size() / // sizeof(int16_t)" and so on when this fix in audio_processing has made @@ -168,8 +98,11 @@ int do_main(int argc, char* argv[]) { WriteIntData(reinterpret_cast(msg.data().data()), num_reverse_channels * reverse_samples_per_channel, reverse_wav_file.get(), - reverse_pcm_file.get()); + reverse_raw_file.get()); } else if (msg.channel_size() > 0) { + if (FLAGS_raw && !reverse_raw_file) { + reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float")); + } scoped_ptr data(new const float*[num_reverse_channels]); for (int i = 0; i < num_reverse_channels; ++i) { data[i] = reinterpret_cast(msg.channel(i).data()); @@ -178,7 +111,7 @@ int do_main(int argc, char* argv[]) { reverse_samples_per_channel, num_reverse_channels, reverse_wav_file.get(), - reverse_pcm_file.get()); + reverse_raw_file.get()); } } else if (event_msg.type() == Event::STREAM) { frame_count++; @@ -189,11 +122,17 @@ int do_main(int argc, char* argv[]) { const Stream msg = event_msg.stream(); if (msg.has_input_data()) { + if (FLAGS_raw && !input_raw_file) { + input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm")); + } WriteIntData(reinterpret_cast(msg.input_data().data()), num_input_channels * input_samples_per_channel, input_wav_file.get(), - input_pcm_file.get()); + input_raw_file.get()); } else if (msg.input_channel_size() > 0) { + if (FLAGS_raw && !input_raw_file) { + input_raw_file.reset(new RawFile(FLAGS_input_file + ".float")); + } scoped_ptr data(new const float*[num_input_channels]); for (int i = 0; i < num_input_channels; ++i) { data[i] = reinterpret_cast(msg.input_channel(i).data()); @@ -202,15 +141,21 @@ int do_main(int argc, char* argv[]) { input_samples_per_channel, num_input_channels, input_wav_file.get(), - input_pcm_file.get()); + input_raw_file.get()); } if (msg.has_output_data()) { + if (FLAGS_raw && !output_raw_file) { + output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm")); + } WriteIntData(reinterpret_cast(msg.output_data().data()), num_output_channels * output_samples_per_channel, output_wav_file.get(), - output_pcm_file.get()); + output_raw_file.get()); } else if (msg.output_channel_size() > 0) { + if (FLAGS_raw && !output_raw_file) { + output_raw_file.reset(new RawFile(FLAGS_output_file + ".float")); + } scoped_ptr data(new const float*[num_output_channels]); for (int i = 0; i < num_output_channels; ++i) { data[i] = @@ -220,7 +165,7 @@ int do_main(int argc, char* argv[]) { output_samples_per_channel, num_output_channels, output_wav_file.get(), - output_pcm_file.get()); + output_raw_file.get()); } if (FLAGS_full) { @@ -287,24 +232,16 @@ int do_main(int argc, char* argv[]) { input_samples_per_channel = input_sample_rate / 100; output_samples_per_channel = output_sample_rate / 100; - if (FLAGS_pcm) { - if (!reverse_pcm_file.get()) { - reverse_pcm_file.reset(new PcmFile(FLAGS_reverse_file)); - } - if (!input_pcm_file.get()) { - input_pcm_file.reset(new PcmFile(FLAGS_input_file)); - } - if (!output_pcm_file.get()) { - output_pcm_file.reset(new PcmFile(FLAGS_output_file)); - } - } else { - reverse_wav_file.reset(new WavFile(FLAGS_reverse_wav_file, + if (!FLAGS_raw) { + // The WAV files need to be reset every time, because they cant change + // their sample rate or number of channels. + reverse_wav_file.reset(new WavFile(FLAGS_reverse_file + ".wav", reverse_sample_rate, num_reverse_channels)); - input_wav_file.reset(new WavFile(FLAGS_input_wav_file, + input_wav_file.reset(new WavFile(FLAGS_input_file + ".wav", input_sample_rate, num_input_channels)); - output_wav_file.reset(new WavFile(FLAGS_output_wav_file, + output_wav_file.reset(new WavFile(FLAGS_output_file + ".wav", output_sample_rate, num_output_channels)); }