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ffmpeg: Properly handle non-planar formats 

For non-planar formats, only the first data plane is used. Therefore,
they need to be handled differently in certain places.
This commit is contained in:
zhupengfei 2020-02-27 16:37:06 +08:00
parent c9c26955d2
commit a28eac08ae
No known key found for this signature in database
GPG Key ID: DD129E108BD09378
1 changed files with 15 additions and 18 deletions
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27
src/core/dumping/ffmpeg_backend.cpp
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@ -227,20 +227,7 @@ bool FFmpegAudioStream::Init(AVFormatContext* format_context) {
codec_context->codec_type = AVMEDIA_TYPE_AUDIO;
codec_context->bit_rate = Settings::values.audio_bitrate;
if (codec->sample_fmts) {
codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
// Use any planar format
const AVSampleFormat* ptr = codec->sample_fmts;
while ((*ptr) != -1) {
if (av_sample_fmt_is_planar((*ptr))) {
codec_context->sample_fmt = (*ptr);
break;
}
ptr++;
}
if (codec_context->sample_fmt == AV_SAMPLE_FMT_NONE) {
LOG_ERROR(Render, "Specified audio encoder does not support any planar format");
return false;
}
codec_context->sample_fmt = codec->sample_fmts[0];
} else {
codec_context->sample_fmt = AV_SAMPLE_FMT_S16P;
}
@ -341,8 +328,14 @@ void FFmpegAudioStream::ProcessFrame(const VariableAudioFrame& channel0,
const auto sample_size = av_get_bytes_per_sample(codec_context->sample_fmt);
std::array<const u8*, 2> src_data = {reinterpret_cast<const u8*>(channel0.data()),
reinterpret_cast<const u8*>(channel1.data())};
std::array<u8*, 2> dst_data = {resampled_data[0] + sample_size * offset,
std::array<u8*, 2> dst_data;
if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
dst_data = {resampled_data[0] + sample_size * offset,
resampled_data[1] + sample_size * offset};
} else {
dst_data = {resampled_data[0] + sample_size * offset * 2}; // 2 channels
}
auto resampled_count = swr_convert(swr_context.get(), dst_data.data(), frame_size - offset,
src_data.data(), channel0.size());
@ -360,7 +353,9 @@ void FFmpegAudioStream::ProcessFrame(const VariableAudioFrame& channel0,
// Prepare frame
audio_frame->nb_samples = frame_size;
audio_frame->data[0] = resampled_data[0];
if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
audio_frame->data[1] = resampled_data[1];
}
audio_frame->pts = frame_count * frame_size;
frame_count++;
@ -383,7 +378,9 @@ void FFmpegAudioStream::Flush() {
// Send the last samples
audio_frame->nb_samples = offset;
audio_frame->data[0] = resampled_data[0];
if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
audio_frame->data[1] = resampled_data[1];
}
audio_frame->pts = frame_count * frame_size;
SendFrame(audio_frame.get());