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Fixed a really huge audio issue, related to audio timestamps, gaps, and also mono vs stereo issues. This is the best audio support I've accomplished thus far. =)

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This commit is contained in:
Jonathan Thomas
2012-12-03 04:51:17 -06:00
parent 8606ec2a6e
commit e264d4fcab
5 changed files with 109 additions and 25 deletions
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54
src/FFmpegReader.cpp
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@@ -6,7 +6,8 @@ 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) {
seek_audio_frame_found(-1), seek_video_frame_found(-1), resampleCtx(NULL), prev_samples(0), prev_pts(0),
pts_total(0), pts_counter(0), display_debug(false) {
// Init FileInfo struct (clear all values)
InitFileInfo();
@@ -722,11 +723,10 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
while (my_packet->size > 0) {
// re-initialize buffer size (it gets changed in the avcodec_decode_audio2 method call)
int buf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE;
int used = avcodec_decode_audio3(aCodecCtx, audio_buf, &buf_size, my_packet);
int used = avcodec_decode_audio3(aCodecCtx, (short *)audio_buf, &buf_size, my_packet);
if (used < 0) {
// Throw exception
my_packet->size = 0;
throw ErrorDecodingAudio("Error decoding audio samples", target_frame);
}
@@ -739,7 +739,23 @@ 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 = round(packet_samples / info.channels) - 1; // Adjust for zero based array
int pts_remaining_samples = packet_samples / info.channels; // Adjust for zero based array
// DEBUG (FOR AUDIO ISSUES) - Get the audio packet start time (in seconds)
int adjusted_pts = packet->pts + audio_pts_offset;
double audio_seconds = double(adjusted_pts) * info.audio_timebase.ToDouble();
double sample_seconds = float(pts_total) / info.sample_rate;
if (display_debug)
cout << pts_counter << ") PTS: " << adjusted_pts << ", Offset: " << audio_pts_offset << ", PTS Diff: " << (adjusted_pts - prev_pts) << ", Samples: " << pts_remaining_samples << ", Sample PTS ratio: " << (float(adjusted_pts - prev_pts) / pts_remaining_samples) << ", Sample Diff: " << (pts_remaining_samples - prev_samples) << ", Total: " << pts_total << ", PTS Seconds: " << audio_seconds << ", Sample Seconds: " << sample_seconds << ", Seconds Diff: " << (audio_seconds - sample_seconds) << ", raw samples: " << packet_samples << endl;
// DEBUG (FOR AUDIO ISSUES)
prev_pts = adjusted_pts;
pts_total += pts_remaining_samples;
pts_counter++;
prev_samples = pts_remaining_samples;
while (pts_remaining_samples)
{
// Get Samples per frame (for this frame number)
@@ -784,8 +800,8 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
// Audio needs to be converted
// Create an audio resample context object (used to convert audio samples)
if (!resampleCtx)
resampleCtx = av_audio_resample_init(
//if (!resampleCtx)
ReSampleContext *resampleCtx = av_audio_resample_init(
info.channels,
info.channels,
info.sample_rate,
@@ -808,6 +824,13 @@ void FFmpegReader::ProcessAudioPacket(int requested_frame, int target_frame, int
delete[] converted_audio;
converted_audio = NULL;
}
// Close audio resample context
if (resampleCtx)
{
audio_resample_close(resampleCtx);
resampleCtx = NULL;
}
}
int starting_frame_number = -1;
@@ -815,7 +838,7 @@ 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 = round(packet_samples / info.channels);
int channel_buffer_size = packet_samples / info.channels;
float *channel_buffer = new float[channel_buffer_size];
// Init buffer array
@@ -1109,19 +1132,28 @@ audio_packet_location FFmpegReader::GetAudioPTSLocation(int pts)
// Calculate the sample # to start on
int sample_start = round(double(samples_per_frame) * sample_start_percentage);
// Protect against broken (i.e. negative) timestamps
if (whole_frame < 1)
whole_frame = 1;
if (sample_start < 0)
sample_start = 0;
// Prepare final audio packet location
audio_packet_location location = {whole_frame, sample_start};
// Compare to previous audio packet (and fix small gaps due to varying PTS timestamps)
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.is_near(previous_packet_location, samples_per_frame, 1000))
{
int orig_frame = location.frame;
int orig_start = location.sample_start;
// Update sample start, to prevent gaps in audio
if (previous_packet_location.sample_start + 1 <= samples_per_frame)
location.sample_start = previous_packet_location.sample_start + 1;
{
location.sample_start = previous_packet_location.sample_start;
location.frame = previous_packet_location.frame;
}
else
{
// set to next frame (since we exceeded the # of samples on a frame)
@@ -1131,6 +1163,8 @@ audio_packet_location FFmpegReader::GetAudioPTSLocation(int pts)
//cout << "GAP DETECTED!!! Changing frame " << orig_frame << ":" << orig_start << " to frame " << location.frame << ":" << location.sample_start << endl;
}
//else
// cout << "NOT NEAR!!! frame " << location.frame << ":" << location.sample_start << " prev frame " << previous_packet_location.frame << ":" << previous_packet_location.sample_start << endl;
// Set previous location
previous_packet_location = location;