雾聪
2023-12-15 3313eb681e34d34292019dc20f6a1aff48a6dcfc
fi bug of FfmpegLoad
2个文件已修改
40 ■■■■■ 已修改文件
runtime/onnxruntime/include/audio.h 4 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
runtime/onnxruntime/src/audio.cpp 36 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
runtime/onnxruntime/include/audio.h
@@ -61,10 +61,10 @@
    void Disp();
    void WavResample(int32_t sampling_rate, const float *waveform, int32_t n);
    bool LoadWav(const char* buf, int n_len, int32_t* sampling_rate);
    bool LoadWav(const char* filename, int32_t* sampling_rate);
    bool LoadWav(const char* filename, int32_t* sampling_rate, bool resample=true);
    bool LoadWav2Char(const char* filename, int32_t* sampling_rate);
    bool LoadPcmwav(const char* buf, int n_file_len, int32_t* sampling_rate);
    bool LoadPcmwav(const char* filename, int32_t* sampling_rate);
    bool LoadPcmwav(const char* filename, int32_t* sampling_rate, bool resample=true);
    bool LoadPcmwav2Char(const char* filename, int32_t* sampling_rate);
    bool LoadOthers2Char(const char* filename);
    bool FfmpegLoad(const char *filename, bool copy2char=false);
runtime/onnxruntime/src/audio.cpp
@@ -354,9 +354,7 @@
                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,
                    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);
@@ -364,20 +362,20 @@
                    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);
                }
            }
        }
@@ -539,9 +537,7 @@
                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,
                    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);
@@ -549,20 +545,20 @@
                    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);
                }
            }
        }
@@ -614,7 +610,7 @@
}
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) {
@@ -676,7 +672,7 @@
        }
        //resample
        if(*sampling_rate != dest_sample_rate){
        if(resample && *sampling_rate != dest_sample_rate){
            WavResample(*sampling_rate, speech_data, speech_len);
        }
@@ -867,7 +863,7 @@
        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) {
        free(speech_data);
@@ -908,7 +904,7 @@
        }
        //resample
        if(*sampling_rate != dest_sample_rate){
        if(resample && *sampling_rate != dest_sample_rate){
            WavResample(*sampling_rate, speech_data, speech_len);
        }