From 28ccfbfc51068a663a80764e14074df5edf2b5ba Mon Sep 17 00:00:00 2001
From: kongdeqiang <kongdeqiang960204@163.com>
Date: 星期五, 13 三月 2026 17:41:41 +0800
Subject: [PATCH] 提交
---
runtime/onnxruntime/src/audio.cpp | 484 +++++++++++++++++++++++++++++++++++------------------
1 files changed, 321 insertions(+), 163 deletions(-)
diff --git a/runtime/onnxruntime/src/audio.cpp b/runtime/onnxruntime/src/audio.cpp
index b543797..22a9ecd 100644
--- a/runtime/onnxruntime/src/audio.cpp
+++ b/runtime/onnxruntime/src/audio.cpp
@@ -133,6 +133,7 @@
};
~AudioWindow(){
free(window);
+ window = nullptr;
};
int put(int val)
{
@@ -160,8 +161,9 @@
len = end - start;
}
AudioFrame::~AudioFrame(){
- if(data != NULL){
+ if(data != nullptr){
free(data);
+ data = nullptr;
}
}
int AudioFrame::SetStart(int val)
@@ -193,63 +195,88 @@
return 0;
}
-Audio::Audio(int data_type) : data_type(data_type)
+Audio::Audio(int data_type) : dest_sample_rate(MODEL_SAMPLE_RATE), data_type(data_type)
{
- speech_buff = NULL;
- speech_data = NULL;
+ speech_buff = nullptr;
+ speech_data = nullptr;
align_size = 1360;
+ seg_sample = dest_sample_rate / 1000;
}
-Audio::Audio(int data_type, int size) : data_type(data_type)
+Audio::Audio(int model_sample_rate, int data_type) : dest_sample_rate(model_sample_rate), data_type(data_type)
{
- speech_buff = NULL;
- speech_data = NULL;
+ speech_buff = nullptr;
+ speech_data = nullptr;
+ align_size = 1360;
+ seg_sample = dest_sample_rate / 1000;
+}
+
+Audio::Audio(int model_sample_rate, int data_type, int size) : dest_sample_rate(model_sample_rate), data_type(data_type)
+{
+ speech_buff = nullptr;
+ speech_data = nullptr;
align_size = (float)size;
+ seg_sample = dest_sample_rate / 1000;
}
Audio::~Audio()
{
- if (speech_buff != NULL) {
+ if (speech_buff != nullptr) {
free(speech_buff);
+ speech_buff = nullptr;
}
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_char != NULL) {
+ if (speech_char != nullptr) {
free(speech_char);
+ speech_char = nullptr;
+ }
+ ClearQueue(frame_queue);
+ ClearQueue(asr_online_queue);
+ ClearQueue(asr_offline_queue);
+}
+
+void Audio::ClearQueue(std::queue<AudioFrame*>& q) {
+ while (!q.empty()) {
+ AudioFrame* frame = q.front();
+ delete frame;
+ q.pop();
}
}
void Audio::Disp()
{
- LOG(INFO) << "Audio time is " << (float)speech_len / MODEL_SAMPLE_RATE << " s. len is " << speech_len;
+ LOG(INFO) << "Audio time is " << (float)speech_len / dest_sample_rate << " s. len is " << speech_len;
}
float Audio::GetTimeLen()
{
- return (float)speech_len / MODEL_SAMPLE_RATE;
+ return (float)speech_len / dest_sample_rate;
}
void Audio::WavResample(int32_t sampling_rate, const float *waveform,
int32_t n)
{
- LOG(INFO) << "Creating a resampler:\n"
- << " in_sample_rate: "<< sampling_rate << "\n"
- << " output_sample_rate: " << static_cast<int32_t>(MODEL_SAMPLE_RATE);
+ LOG(INFO) << "Creating a resampler: "
+ << " in_sample_rate: "<< sampling_rate
+ << " output_sample_rate: " << static_cast<int32_t>(dest_sample_rate);
float min_freq =
- std::min<int32_t>(sampling_rate, MODEL_SAMPLE_RATE);
+ std::min<int32_t>(sampling_rate, dest_sample_rate);
float lowpass_cutoff = 0.99 * 0.5 * min_freq;
int32_t lowpass_filter_width = 6;
auto resampler = std::make_unique<LinearResample>(
- sampling_rate, MODEL_SAMPLE_RATE, lowpass_cutoff, lowpass_filter_width);
+ sampling_rate, dest_sample_rate, lowpass_cutoff, lowpass_filter_width);
std::vector<float> samples;
resampler->Resample(waveform, n, true, &samples);
//reset speech_data
speech_len = samples.size();
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
speech_data = (float*)malloc(sizeof(float) * speech_len);
memset(speech_data, 0, sizeof(float) * speech_len);
@@ -262,21 +289,21 @@
#else
// from file
AVFormatContext* formatContext = avformat_alloc_context();
- if (avformat_open_input(&formatContext, filename, NULL, NULL) != 0) {
+ if (avformat_open_input(&formatContext, filename, nullptr, nullptr) != 0) {
LOG(ERROR) << "Error: Could not open input file.";
avformat_close_input(&formatContext);
avformat_free_context(formatContext);
return false;
}
- if (avformat_find_stream_info(formatContext, NULL) < 0) {
+ if (avformat_find_stream_info(formatContext, nullptr) < 0) {
LOG(ERROR) << "Error: Could not open input file.";
avformat_close_input(&formatContext);
avformat_free_context(formatContext);
return false;
}
- const AVCodec* codec = NULL;
- AVCodecParameters* codecParameters = NULL;
+ const AVCodec* codec = nullptr;
+ AVCodecParameters* codecParameters = nullptr;
int audioStreamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (audioStreamIndex >= 0) {
codecParameters = formatContext->streams[audioStreamIndex]->codecpar;
@@ -300,7 +327,7 @@
avcodec_free_context(&codecContext);
return false;
}
- if (avcodec_open2(codecContext, codec, NULL) < 0) {
+ if (avcodec_open2(codecContext, codec, nullptr) < 0) {
LOG(ERROR) << "Error: Could not open audio decoder.";
avformat_close_input(&formatContext);
avformat_free_context(formatContext);
@@ -311,7 +338,7 @@
nullptr, // allocate a new context
AV_CH_LAYOUT_MONO, // output channel layout (stereo)
AV_SAMPLE_FMT_S16, // output sample format (signed 16-bit)
- 16000, // output sample rate (same as input)
+ dest_sample_rate, // output sample rate (same as input)
av_get_default_channel_layout(codecContext->channels), // input channel layout
codecContext->sample_fmt, // input sample format
codecContext->sample_rate, // input sample rate
@@ -344,30 +371,28 @@
while (avcodec_receive_frame(codecContext, frame) >= 0) {
// Resample audio if necessary
std::vector<uint8_t> resampled_buffer;
- int in_samples = frame->nb_samples;
- uint8_t **in_data = frame->extended_data;
- int out_samples = av_rescale_rnd(in_samples,
- 16000,
+ int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, codecContext->sample_rate) + frame->nb_samples,
+ dest_sample_rate,
codecContext->sample_rate,
AV_ROUND_DOWN);
int resampled_size = out_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
if (resampled_buffer.size() < resampled_size) {
resampled_buffer.resize(resampled_size);
- }
+ }
uint8_t *resampled_data = resampled_buffer.data();
int ret = swr_convert(
swr_ctx,
&resampled_data, // output buffer
- resampled_size, // output buffer size
- (const uint8_t **)(frame->data), //(const uint8_t **)(frame->extended_data)
- in_samples // input buffer size
+ out_samples, // output buffer size
+ (const uint8_t **)(frame->data), // choose channel
+ frame->nb_samples // input buffer size
);
if (ret < 0) {
LOG(ERROR) << "Error resampling audio";
break;
}
- std::copy(resampled_buffer.begin(), resampled_buffer.end(), std::back_inserter(resampled_buffers));
+ resampled_buffers.insert(resampled_buffers.end(), resampled_buffer.begin(), resampled_buffer.begin() + resampled_size);
}
}
}
@@ -381,14 +406,13 @@
av_packet_free(&packet);
av_frame_free(&frame);
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
- free(speech_buff);
- }
- if (speech_char != NULL) {
+ if (speech_char != nullptr) {
free(speech_char);
+ speech_char = nullptr;
}
offset = 0;
@@ -399,30 +423,25 @@
}
speech_len = (resampled_buffers.size()) / 2;
- speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
- if (speech_buff)
- {
- memset(speech_buff, 0, sizeof(int16_t) * speech_len);
- memcpy((void*)speech_buff, (const void*)resampled_buffers.data(), speech_len * sizeof(int16_t));
-
- speech_data = (float*)malloc(sizeof(float) * speech_len);
+ speech_data = (float*)malloc(sizeof(float) * speech_len);
+ if(speech_data){
memset(speech_data, 0, sizeof(float) * speech_len);
-
float scale = 1;
if (data_type == 1) {
- scale = 32768;
+ scale = 32768.0f;
}
- for (int32_t i = 0; i != speech_len; ++i) {
- speech_data[i] = (float)speech_buff[i] / scale;
+ for (int32_t i = 0; i < speech_len; ++i) {
+ int16_t val = (int16_t)((resampled_buffers[2 * i + 1] << 8) | resampled_buffers[2 * i]);
+ speech_data[i] = (float)val / scale;
}
-
AudioFrame* frame = new AudioFrame(speech_len);
frame_queue.push(frame);
return true;
- }
- else
+ }else{
return false;
+ }
+
#endif
}
@@ -443,9 +462,13 @@
nullptr, // write callback (not used here)
nullptr // seek callback (not used here)
);
+ if (!avio_ctx) {
+ av_free(buf_copy);
+ return false;
+ }
AVFormatContext* formatContext = avformat_alloc_context();
formatContext->pb = avio_ctx;
- if (avformat_open_input(&formatContext, "", NULL, NULL) != 0) {
+ if (avformat_open_input(&formatContext, "", nullptr, nullptr) != 0) {
LOG(ERROR) << "Error: Could not open input file.";
avio_context_free(&avio_ctx);
avformat_close_input(&formatContext);
@@ -453,15 +476,15 @@
return false;
}
- if (avformat_find_stream_info(formatContext, NULL) < 0) {
+ if (avformat_find_stream_info(formatContext, nullptr) < 0) {
LOG(ERROR) << "Error: Could not find stream information.";
avio_context_free(&avio_ctx);
avformat_close_input(&formatContext);
avformat_free_context(formatContext);
return false;
}
- const AVCodec* codec = NULL;
- AVCodecParameters* codecParameters = NULL;
+ const AVCodec* codec = nullptr;
+ AVCodecParameters* codecParameters = nullptr;
int audioStreamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (audioStreamIndex >= 0) {
codecParameters = formatContext->streams[audioStreamIndex]->codecpar;
@@ -482,7 +505,7 @@
avcodec_free_context(&codecContext);
return false;
}
- if (avcodec_open2(codecContext, codec, NULL) < 0) {
+ if (avcodec_open2(codecContext, codec, nullptr) < 0) {
LOG(ERROR) << "Error: Could not open audio decoder.";
avio_context_free(&avio_ctx);
avformat_close_input(&formatContext);
@@ -494,7 +517,7 @@
nullptr, // allocate a new context
AV_CH_LAYOUT_MONO, // output channel layout (stereo)
AV_SAMPLE_FMT_S16, // output sample format (signed 16-bit)
- 16000, // output sample rate (same as input)
+ dest_sample_rate, // output sample rate (same as input)
av_get_default_channel_layout(codecContext->channels), // input channel layout
codecContext->sample_fmt, // input sample format
codecContext->sample_rate, // input sample rate
@@ -529,37 +552,37 @@
while (avcodec_receive_frame(codecContext, frame) >= 0) {
// Resample audio if necessary
std::vector<uint8_t> resampled_buffer;
- int in_samples = frame->nb_samples;
- uint8_t **in_data = frame->extended_data;
- int out_samples = av_rescale_rnd(in_samples,
- 16000,
+ int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, codecContext->sample_rate) + frame->nb_samples,
+ dest_sample_rate,
codecContext->sample_rate,
AV_ROUND_DOWN);
int resampled_size = out_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
if (resampled_buffer.size() < resampled_size) {
resampled_buffer.resize(resampled_size);
- }
+ }
uint8_t *resampled_data = resampled_buffer.data();
int ret = swr_convert(
swr_ctx,
&resampled_data, // output buffer
- resampled_size, // output buffer size
- (const uint8_t **)(frame->data), //(const uint8_t **)(frame->extended_data)
- in_samples // input buffer size
+ out_samples, // output buffer size
+ (const uint8_t **)(frame->data), // choose channel: channel_data
+ frame->nb_samples // input buffer size
);
if (ret < 0) {
LOG(ERROR) << "Error resampling audio";
break;
}
- std::copy(resampled_buffer.begin(), resampled_buffer.end(), std::back_inserter(resampled_buffers));
+ resampled_buffers.insert(resampled_buffers.end(), resampled_buffer.begin(), resampled_buffer.begin() + resampled_size);
}
}
}
av_packet_unref(packet);
}
- avio_context_free(&avio_ctx);
+ //avio_context_free(&avio_ctx);
+ av_freep(&avio_ctx ->buffer);
+ av_freep(&avio_ctx);
avformat_close_input(&formatContext);
avformat_free_context(formatContext);
avcodec_free_context(&codecContext);
@@ -567,51 +590,45 @@
av_packet_free(&packet);
av_frame_free(&frame);
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
- free(speech_buff);
- }
- offset = 0;
speech_len = (resampled_buffers.size()) / 2;
- speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
- if (speech_buff)
- {
- memset(speech_buff, 0, sizeof(int16_t) * speech_len);
- memcpy((void*)speech_buff, (const void*)resampled_buffers.data(), speech_len * sizeof(int16_t));
-
- speech_data = (float*)malloc(sizeof(float) * speech_len);
+ speech_data = (float*)malloc(sizeof(float) * speech_len);
+ if(speech_data){
memset(speech_data, 0, sizeof(float) * speech_len);
-
float scale = 1;
if (data_type == 1) {
- scale = 32768;
+ scale = 32768.0f;
}
- for (int32_t i = 0; i != speech_len; ++i) {
- speech_data[i] = (float)speech_buff[i] / scale;
+ for (int32_t i = 0; i < speech_len; ++i) {
+ int16_t val = (int16_t)((resampled_buffers[2 * i + 1] << 8) | resampled_buffers[2 * i]);
+ speech_data[i] = (float)val / scale;
}
-
AudioFrame* frame = new AudioFrame(speech_len);
frame_queue.push(frame);
return true;
- }
- else
+ }else{
return false;
+ }
+
#endif
}
-bool Audio::LoadWav(const char *filename, int32_t* sampling_rate)
+bool Audio::LoadWav(const char *filename, int32_t* sampling_rate, bool resample)
{
WaveHeader header;
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
+ if (speech_buff != nullptr) {
free(speech_buff);
+ speech_buff = nullptr;
}
offset = 0;
@@ -666,7 +683,7 @@
}
//resample
- if(*sampling_rate != MODEL_SAMPLE_RATE){
+ if(resample && *sampling_rate != dest_sample_rate){
WavResample(*sampling_rate, speech_data, speech_len);
}
@@ -682,8 +699,9 @@
bool Audio::LoadWav2Char(const char *filename, int32_t* sampling_rate)
{
WaveHeader header;
- if (speech_char != NULL) {
+ if (speech_char != nullptr) {
free(speech_char);
+ speech_char = nullptr;
}
offset = 0;
std::ifstream is(filename, std::ifstream::binary);
@@ -721,13 +739,14 @@
bool Audio::LoadWav(const char* buf, int n_file_len, int32_t* sampling_rate)
{
WaveHeader header;
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
+ if (speech_buff != nullptr) {
free(speech_buff);
+ speech_buff = nullptr;
}
- offset = 0;
std::memcpy(&header, buf, sizeof(header));
@@ -752,7 +771,7 @@
}
//resample
- if(*sampling_rate != MODEL_SAMPLE_RATE){
+ if(*sampling_rate != dest_sample_rate){
WavResample(*sampling_rate, speech_data, speech_len);
}
@@ -767,80 +786,60 @@
bool Audio::LoadPcmwav(const char* buf, int n_buf_len, int32_t* sampling_rate)
{
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
- free(speech_buff);
- }
- offset = 0;
speech_len = n_buf_len / 2;
- speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
- if (speech_buff)
- {
- memset(speech_buff, 0, sizeof(int16_t) * speech_len);
- memcpy((void*)speech_buff, (const void*)buf, speech_len * sizeof(int16_t));
-
- speech_data = (float*)malloc(sizeof(float) * speech_len);
- memset(speech_data, 0, sizeof(float) * speech_len);
-
+ speech_data = (float*)malloc(sizeof(float) * speech_len);
+ if(speech_data){
float scale = 1;
if (data_type == 1) {
- scale = 32768;
+ scale = 32768.0f;
+ }
+ const uint8_t* byte_buf = reinterpret_cast<const uint8_t*>(buf);
+ for (int32_t i = 0; i < speech_len; ++i) {
+ int16_t val = (int16_t)((byte_buf[2 * i + 1] << 8) | byte_buf[2 * i]);
+ speech_data[i] = (float)val / scale;
}
- for (int32_t i = 0; i != speech_len; ++i) {
- speech_data[i] = (float)speech_buff[i] / scale;
- }
-
//resample
- if(*sampling_rate != MODEL_SAMPLE_RATE){
+ if(*sampling_rate != dest_sample_rate){
WavResample(*sampling_rate, speech_data, speech_len);
}
AudioFrame* frame = new AudioFrame(speech_len);
frame_queue.push(frame);
+
return true;
-
- }
- else
+ }else{
return false;
+ }
}
bool Audio::LoadPcmwavOnline(const char* buf, int n_buf_len, int32_t* sampling_rate)
{
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
- }
- if (speech_buff != NULL) {
- free(speech_buff);
- }
- if (speech_char != NULL) {
- free(speech_char);
+ speech_data = nullptr;
}
speech_len = n_buf_len / 2;
- speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
- if (speech_buff)
- {
- memset(speech_buff, 0, sizeof(int16_t) * speech_len);
- memcpy((void*)speech_buff, (const void*)buf, speech_len * sizeof(int16_t));
-
- speech_data = (float*)malloc(sizeof(float) * speech_len);
- memset(speech_data, 0, sizeof(float) * speech_len);
-
+ speech_data = (float*)malloc(sizeof(float) * speech_len);
+ if(speech_data){
float scale = 1;
if (data_type == 1) {
- scale = 32768;
+ scale = 32768.0f;
+ }
+ const uint8_t* byte_buf = reinterpret_cast<const uint8_t*>(buf);
+ for (int32_t i = 0; i < speech_len; ++i) {
+ int16_t val = (int16_t)((byte_buf[2 * i + 1] << 8) | byte_buf[2 * i]);
+ speech_data[i] = (float)val / scale;
}
- for (int32_t i = 0; i != speech_len; ++i) {
- speech_data[i] = (float)speech_buff[i] / scale;
- }
-
//resample
- if(*sampling_rate != MODEL_SAMPLE_RATE){
+ if(*sampling_rate != dest_sample_rate){
WavResample(*sampling_rate, speech_data, speech_len);
}
@@ -850,20 +849,22 @@
AudioFrame* frame = new AudioFrame(speech_len);
frame_queue.push(frame);
+
return true;
-
- }
- else
+ }else{
return false;
+ }
}
-bool Audio::LoadPcmwav(const char* filename, int32_t* sampling_rate)
+bool Audio::LoadPcmwav(const char* filename, int32_t* sampling_rate, bool resample)
{
- if (speech_data != NULL) {
+ if (speech_data != nullptr) {
free(speech_data);
+ speech_data = nullptr;
}
- if (speech_buff != NULL) {
+ if (speech_buff != nullptr) {
free(speech_buff);
+ speech_buff = nullptr;
}
offset = 0;
@@ -898,7 +899,7 @@
}
//resample
- if(*sampling_rate != MODEL_SAMPLE_RATE){
+ if(resample && *sampling_rate != dest_sample_rate){
WavResample(*sampling_rate, speech_data, speech_len);
}
@@ -914,8 +915,9 @@
bool Audio::LoadPcmwav2Char(const char* filename, int32_t* sampling_rate)
{
- if (speech_char != NULL) {
+ if (speech_char != nullptr) {
free(speech_char);
+ speech_char = nullptr;
}
offset = 0;
@@ -941,8 +943,9 @@
bool Audio::LoadOthers2Char(const char* filename)
{
- if (speech_char != NULL) {
+ if (speech_char != nullptr) {
free(speech_char);
+ speech_char = nullptr;
}
FILE* fp;
@@ -1009,7 +1012,7 @@
AudioFrame *frame = frame_queue.front();
frame_queue.pop();
- start_time = (float)(frame->GetStart())/MODEL_SAMPLE_RATE;
+ start_time = (float)(frame->GetStart())/ dest_sample_rate;
dout = speech_data + frame->GetStart();
len = frame->GetLen();
delete frame;
@@ -1017,6 +1020,90 @@
return 1;
} else {
return 0;
+ }
+}
+
+int Audio::Fetch(float**& dout, int*& len, int*& flag, float*& start_time, int batch_size, int &batch_in)
+{
+ batch_in = std::min((int)frame_queue.size(), batch_size);
+ if (batch_in == 0){
+ return 0;
+ } else{
+ // init
+ dout = new float*[batch_in];
+ len = new int[batch_in];
+ flag = new int[batch_in];
+ start_time = new float[batch_in];
+
+ for(int idx=0; idx < batch_in; idx++){
+ AudioFrame *frame = frame_queue.front();
+ frame_queue.pop();
+
+ start_time[idx] = (float)(frame->GetStart())/ dest_sample_rate;
+ dout[idx] = speech_data + frame->GetStart();
+ len[idx] = frame->GetLen();
+ delete frame;
+ flag[idx] = S_END;
+ }
+ return 1;
+ }
+}
+
+int Audio::FetchDynamic(float**& dout, int*& len, int*& flag, float*& start_time, int batch_size, int &batch_in)
+{
+ //compute batch size
+ queue<AudioFrame *> frame_batch;
+ int max_acc = 300*1000*seg_sample;
+ int max_sent = 60*1000*seg_sample;
+ int bs_acc = 0;
+ int max_len = 0;
+ int max_batch = 1;
+ #ifdef USE_GPU
+ max_batch = batch_size;
+ #endif
+ max_batch = std::min(max_batch, (int)frame_queue.size());
+
+ for(int idx=0; idx < max_batch; idx++){
+ AudioFrame *frame = frame_queue.front();
+ int length = frame->GetLen();
+ if(length >= max_sent){
+ if(bs_acc == 0){
+ bs_acc++;
+ frame_batch.push(frame);
+ frame_queue.pop();
+ }
+ break;
+ }
+ max_len = std::max(max_len, frame->GetLen());
+ if(max_len*(bs_acc+1) > max_acc){
+ break;
+ }
+ bs_acc++;
+ frame_batch.push(frame);
+ frame_queue.pop();
+ }
+
+ batch_in = (int)frame_batch.size();
+ if (batch_in == 0){
+ return 0;
+ } else{
+ // init
+ dout = new float*[batch_in];
+ len = new int[batch_in];
+ flag = new int[batch_in];
+ start_time = new float[batch_in];
+
+ for(int idx=0; idx < batch_in; idx++){
+ AudioFrame *frame = frame_batch.front();
+ frame_batch.pop();
+
+ start_time[idx] = (float)(frame->GetStart())/ dest_sample_rate;
+ dout[idx] = speech_data + frame->GetStart();
+ len[idx] = frame->GetLen();
+ delete frame;
+ flag[idx] = S_END;
+ }
+ return 1;
}
}
@@ -1047,6 +1134,7 @@
new_data[tmp_off + i] = speech_data[ii];
}
free(speech_data);
+ speech_data = nullptr;
speech_data = new_data;
speech_len = num_new_samples;
@@ -1065,7 +1153,7 @@
frame_queue.pop();
int sp_len = frame->GetLen();
delete frame;
- frame = NULL;
+ frame = nullptr;
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
vector<std::vector<int>> vad_segments = (offline_stream->vad_handle)->Infer(pcm_data);
@@ -1077,7 +1165,77 @@
frame->SetStart(start);
frame->SetEnd(end);
frame_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
+ }
+}
+
+void Audio::CutSplit(OfflineStream* offline_stream, std::vector<int> &index_vector)
+{
+ std::unique_ptr<VadModel> vad_online_handle = make_unique<FsmnVadOnline>((FsmnVad*)(offline_stream->vad_handle).get());
+ AudioFrame *frame;
+
+ frame = frame_queue.front();
+ frame_queue.pop();
+ int sp_len = frame->GetLen();
+ delete frame;
+ frame = nullptr;
+
+ int step = dest_sample_rate*1;
+ bool is_final=false;
+ vector<std::vector<int>> vad_segments;
+ for (int sample_offset = 0; sample_offset < speech_len; sample_offset += std::min(step, speech_len - sample_offset)) {
+ if (sample_offset + step >= speech_len - 1) {
+ step = speech_len - sample_offset;
+ is_final = true;
+ } else {
+ is_final = false;
+ }
+ std::vector<float> pcm_data(speech_data+sample_offset, speech_data+sample_offset+step);
+ vector<std::vector<int>> cut_segments = vad_online_handle->Infer(pcm_data, is_final);
+ vad_segments.insert(vad_segments.end(), cut_segments.begin(), cut_segments.end());
+ }
+
+ int speech_start_i = -1, speech_end_i =-1;
+ std::vector<AudioFrame*> vad_frames;
+ for(vector<int> vad_segment:vad_segments)
+ {
+ if(vad_segment.size() != 2){
+ LOG(ERROR) << "Size of vad_segment is not 2.";
+ break;
+ }
+ if(vad_segment[0] != -1){
+ speech_start_i = vad_segment[0];
+ }
+ if(vad_segment[1] != -1){
+ speech_end_i = vad_segment[1];
+ }
+
+ if(speech_start_i!=-1 && speech_end_i!=-1){
+ int start = speech_start_i*seg_sample;
+ int end = speech_end_i*seg_sample;
+ frame = new AudioFrame(end-start);
+ frame->SetStart(start);
+ frame->SetEnd(end);
+ vad_frames.push_back(frame);
+ frame = nullptr;
+ speech_start_i=-1;
+ speech_end_i=-1;
+ }
+
+ }
+ // sort
+ {
+ index_vector.clear();
+ index_vector.resize(vad_frames.size());
+ for (int i = 0; i < index_vector.size(); ++i) {
+ index_vector[i] = i;
+ }
+ std::sort(index_vector.begin(), index_vector.end(), [&vad_frames](const int a, const int b) {
+ return vad_frames[a]->len < vad_frames[b]->len;
+ });
+ for (int idx : index_vector) {
+ frame_queue.push(vad_frames[idx]);
+ }
}
}
@@ -1089,7 +1247,7 @@
frame_queue.pop();
int sp_len = frame->GetLen();
delete frame;
- frame = NULL;
+ frame = nullptr;
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
vad_segments = vad_obj->Infer(pcm_data, input_finished);
@@ -1104,7 +1262,7 @@
frame_queue.pop();
int sp_len = frame->GetLen();
delete frame;
- frame = NULL;
+ frame = nullptr;
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
vector<std::vector<int>> vad_segments = vad_obj->Infer(pcm_data, input_finished);
@@ -1125,7 +1283,7 @@
frame->data = (float*)malloc(sizeof(float) * step);
memcpy(frame->data, all_samples.data()+start-offset, step*sizeof(float));
asr_online_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
speech_start += step/seg_sample;
}
}
@@ -1153,7 +1311,7 @@
frame->data = (float*)malloc(sizeof(float) * (end-start));
memcpy(frame->data, all_samples.data()+start-offset, (end-start)*sizeof(float));
asr_online_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
}
if(asr_mode != ASR_ONLINE){
@@ -1164,7 +1322,7 @@
frame->data = (float*)malloc(sizeof(float) * (end-start));
memcpy(frame->data, all_samples.data()+start-offset, (end-start)*sizeof(float));
asr_offline_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
}
speech_start = -1;
@@ -1187,7 +1345,7 @@
frame->data = (float*)malloc(sizeof(float) * step);
memcpy(frame->data, all_samples.data()+start-offset, step*sizeof(float));
asr_online_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
speech_start += step/seg_sample;
}
}
@@ -1212,7 +1370,7 @@
frame->data = (float*)malloc(sizeof(float) * (end-offline_start));
memcpy(frame->data, all_samples.data()+offline_start-offset, (end-offline_start)*sizeof(float));
asr_offline_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
}
if(asr_mode != ASR_OFFLINE){
@@ -1230,7 +1388,7 @@
frame->data = (float*)malloc(sizeof(float) * step);
memcpy(frame->data, all_samples.data()+start-offset+sample_offset, step*sizeof(float));
asr_online_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
}
}else{
frame = new AudioFrame(0);
@@ -1238,7 +1396,7 @@
frame->global_start = speech_start; // in this case start >= end
frame->global_end = speech_end_i;
asr_online_queue.push(frame);
- frame = NULL;
+ frame = nullptr;
}
}
speech_start = -1;
@@ -1248,7 +1406,7 @@
}
// erase all_samples
- int vector_cache = MODEL_SAMPLE_RATE*2;
+ int vector_cache = dest_sample_rate*2;
if(speech_offline_start == -1){
if(all_samples.size() > vector_cache){
int erase_num = all_samples.size() - vector_cache;
--
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