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Fixed a big audio bug, where varying timestamp values on low precision audio timebases (i.e. 1/1000) would leave small gaps in the audio wave. Also, fixed a few issues related to sample_rate getting lost between the reader and clip and timeline objects.

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This commit is contained in:
Jonathan Thomas
2012-11-20 16:22:50 -06:00
parent 7e51c204f9
commit 495ca9dc9e
9 changed files with 35 additions and 58 deletions
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41
src/FFmpegReader.cpp
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@@ -6,7 +6,7 @@ FFmpegReader::FFmpegReader(string path) throw(InvalidFile, NoStreamsFound, Inval
: last_frame(0), is_seeking(0), seeking_pts(0), seeking_frame(0), seek_count(0),
audio_pts_offset(99999), video_pts_offset(99999), path(path), is_video_seek(true), check_interlace(false),
check_fps(false), enable_seek(true), rescaler_position(0), num_of_rescalers(32), is_open(false),
seek_audio_frame_found(-1), seek_video_frame_found(-1), resampleCtx(NULL), counter(0), previous_audio_pts(0), total_samples(0) {
seek_audio_frame_found(-1), seek_video_frame_found(-1), resampleCtx(NULL) {
// Init FileInfo struct (clear all values)
InitFileInfo();
@@ -718,12 +718,6 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Allocate audio buffer
int16_t *audio_buf = new int16_t[AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
//if (packet->pts == 4378 || packet->pts == 4402)
// cout << "ProcessAudioPacket: pts: " << packet->pts << ", my_packet->size: " << my_packet->size << endl;
// DEBUG COUNTER
counter++;
int packet_samples = 0;
while (my_packet->size > 0) {
// re-initialize buffer size (it gets changed in the avcodec_decode_audio2 method call)
@@ -745,7 +739,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
}
// Estimate the # of samples and the end of this packet's location (to prevent GAPS for the next timestamp)
int pts_remaining_samples = packet_samples / info.channels;
int pts_remaining_samples = round(packet_samples / info.channels);
while (pts_remaining_samples)
{
// Get Samples per frame (for this frame number)
@@ -770,18 +764,6 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
}
// DEBUG
total_samples += packet_samples;
cout << "packet: " << counter << ", PTS: " << packet->pts << ", PTS Diff: " << (packet->pts - previous_audio_pts) << ", samples: " << packet_samples << ", total: " << total_samples << endl;
previous_audio_pts = packet->pts;
// if (packet->pts >= 4378 && packet->pts <= 4500)
// {
// cout << "ProcessAudioPacket: pts: " << packet->pts << ", packet_samples: " << packet_samples << endl;
// cout << "first samples for pts: " << packet->pts << ", " << audio_buf[0] << ", " << audio_buf[1] << ", " << audio_buf[2] << ", " << audio_buf[3] << ", " << audio_buf[4] << ", " << audio_buf[5] << endl;
// cout << "last samples for pts: " << packet->pts << ", " << audio_buf[2045] << ", " << audio_buf[2046] << ", " << audio_buf[2047] << ", " << audio_buf[2048] << ", " << audio_buf[2049] << ", " << audio_buf[2050] << endl;
// }
#pragma omp critical (packet_cache)
{
// Remove packet
@@ -789,8 +771,6 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
RemoveAVPacket(my_packet);
}
#pragma omp task firstprivate(requested_frame, target_frame, my_cache, starting_sample, audio_buf)
{
@@ -835,7 +815,7 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
{
// Array of floats (to hold samples for each channel)
starting_frame_number = target_frame;
int channel_buffer_size = (packet_samples / info.channels) + 1;
int channel_buffer_size = round(packet_samples / info.channels);
float *channel_buffer = new float[channel_buffer_size];
// Init buffer array
@@ -1136,11 +1116,20 @@ audio_packet_location FFmpegReader::GetAudioPTSLocation(int pts)
int diff_previous_packet = abs(location.sample_start - previous_packet_location.sample_start);
if (location.frame == previous_packet_location.frame && diff_previous_packet >= 0 && diff_previous_packet <= 100)
{
if (location.frame >= 175 && location.frame < 220)
cout << "GAP DETECTED!!! Changing frame " << location.frame << ", sample start: " << location.sample_start << " to " << previous_packet_location.sample_start + 1 << endl;
int orig_frame = location.frame;
int orig_start = location.sample_start;
// Update sample start, to prevent gaps in audio
location.sample_start = previous_packet_location.sample_start + 1;
if (previous_packet_location.sample_start + 1 <= samples_per_frame)
location.sample_start = previous_packet_location.sample_start + 1;
else
{
// set to next frame (since we exceeded the # of samples on a frame)
location.sample_start = 0;
location.frame++;
}
//cout << "GAP DETECTED!!! Changing frame " << orig_frame << ":" << orig_start << " to frame " << location.frame << ":" << location.sample_start << endl;
}
// Set previous location