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Refactored some of the FFmpegReader audio class, trying to locate the pops and cracks in the audio.

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Jonathan Thomas
2012-08-21 02:12:35 -05:00
parent 2b5d317123
commit 0e235b0efb
3 changed files with 33 additions and 35 deletions
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23
src/FFmpegReader.cpp
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@@ -642,22 +642,18 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Init some local variables (for OpenMP)
Cache *my_cache = &working_cache;
AVPacket *my_packet = packets[packet];
int16_t *audio_buf = NULL;
int16_t *converted_audio = NULL;
float *channel_buffer = NULL;
int pts = my_packet->pts;
// Add audio frame to list of processing audio frames
#pragma omp critical (processing_list)
processing_audio_frames[target_frame] = target_frame;
#pragma omp task firstprivate(requested_frame, target_frame, my_cache, my_packet, pts, audio_buf, converted_audio, channel_buffer, starting_sample)
#pragma omp task firstprivate(requested_frame, target_frame, my_cache, my_packet, starting_sample)
{
// Allocate audio buffer
audio_buf = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
int16_t *audio_buf = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
// create a new array (to hold the re-sampled audio)
converted_audio = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
int16_t *converted_audio = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
int packet_samples = 0;
while (my_packet->size > 0) {
@@ -692,6 +688,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Re-sample audio samples (if needed)
if(aCodecCtx->sample_fmt != AV_SAMPLE_FMT_S16) {
// Audio needs to be converted
// Create an audio resample context object (used to convert audio samples)
ReSampleContext *resampleCtx = av_audio_resample_init(
@@ -718,13 +715,19 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
}
}
// DEBUG
//for (int s = 0; s < packet_samples; s++)
// audio_buf[s] = s;
int starting_frame_number = -1;
for (int channel_filter = 0; channel_filter < info.channels; channel_filter++)
{
// Array of floats (to hold samples for each channel)
starting_frame_number = target_frame;
int channel_buffer_size = (packet_samples / info.channels) + 1;
channel_buffer = new float[channel_buffer_size];
float *channel_buffer = new float[channel_buffer_size];
// Init buffer array
for (int z = 0; z < channel_buffer_size; z++)
@@ -810,7 +813,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
delete[] audio_buf;
audio_buf = NULL;
delete[] converted_audio;
converted_audio = 0;
converted_audio = NULL;
// Add video frame to list of processing video frames
#pragma omp critical (processing_list)
@@ -1053,7 +1056,7 @@ Frame* FFmpegReader::CreateFrame(int requested_frame)
int samples_per_frame = GetSamplesPerFrame(requested_frame);
// Create a new frame on the working cache
Frame *f = new Frame(requested_frame, info.width, info.height, "#000000", samples_per_frame, info.channels);
Frame *f = new Frame(requested_frame, 1, 1, "#000000", samples_per_frame, info.channels);
f->SetPixelRatio(info.pixel_ratio.num, info.pixel_ratio.den);
f->SetSampleRate(info.sample_rate);