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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.
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@ -227,20 +227,7 @@ bool FFmpegAudioStream::Init(AVFormatContext* format_context) {
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codec_context->codec_type = AVMEDIA_TYPE_AUDIO;
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codec_context->codec_type = AVMEDIA_TYPE_AUDIO;
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codec_context->bit_rate = Settings::values.audio_bitrate;
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codec_context->bit_rate = Settings::values.audio_bitrate;
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if (codec->sample_fmts) {
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if (codec->sample_fmts) {
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codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
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codec_context->sample_fmt = codec->sample_fmts[0];
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// Use any planar format
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const AVSampleFormat* ptr = codec->sample_fmts;
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while ((*ptr) != -1) {
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if (av_sample_fmt_is_planar((*ptr))) {
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codec_context->sample_fmt = (*ptr);
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break;
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}
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ptr++;
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}
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if (codec_context->sample_fmt == AV_SAMPLE_FMT_NONE) {
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LOG_ERROR(Render, "Specified audio encoder does not support any planar format");
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return false;
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}
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} else {
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} else {
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codec_context->sample_fmt = AV_SAMPLE_FMT_S16P;
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codec_context->sample_fmt = AV_SAMPLE_FMT_S16P;
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}
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}
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@ -341,8 +328,14 @@ void FFmpegAudioStream::ProcessFrame(const VariableAudioFrame& channel0,
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const auto sample_size = av_get_bytes_per_sample(codec_context->sample_fmt);
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const auto sample_size = av_get_bytes_per_sample(codec_context->sample_fmt);
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std::array<const u8*, 2> src_data = {reinterpret_cast<const u8*>(channel0.data()),
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std::array<const u8*, 2> src_data = {reinterpret_cast<const u8*>(channel0.data()),
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reinterpret_cast<const u8*>(channel1.data())};
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reinterpret_cast<const u8*>(channel1.data())};
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std::array<u8*, 2> dst_data = {resampled_data[0] + sample_size * offset,
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resampled_data[1] + sample_size * offset};
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std::array<u8*, 2> dst_data;
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if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
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dst_data = {resampled_data[0] + sample_size * offset,
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resampled_data[1] + sample_size * offset};
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} else {
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dst_data = {resampled_data[0] + sample_size * offset * 2}; // 2 channels
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}
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auto resampled_count = swr_convert(swr_context.get(), dst_data.data(), frame_size - offset,
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auto resampled_count = swr_convert(swr_context.get(), dst_data.data(), frame_size - offset,
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src_data.data(), channel0.size());
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src_data.data(), channel0.size());
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@ -360,7 +353,9 @@ void FFmpegAudioStream::ProcessFrame(const VariableAudioFrame& channel0,
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// Prepare frame
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// Prepare frame
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audio_frame->nb_samples = frame_size;
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audio_frame->nb_samples = frame_size;
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audio_frame->data[0] = resampled_data[0];
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audio_frame->data[0] = resampled_data[0];
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audio_frame->data[1] = resampled_data[1];
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if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
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audio_frame->data[1] = resampled_data[1];
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}
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audio_frame->pts = frame_count * frame_size;
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audio_frame->pts = frame_count * frame_size;
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frame_count++;
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frame_count++;
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@ -383,7 +378,9 @@ void FFmpegAudioStream::Flush() {
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// Send the last samples
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// Send the last samples
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audio_frame->nb_samples = offset;
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audio_frame->nb_samples = offset;
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audio_frame->data[0] = resampled_data[0];
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audio_frame->data[0] = resampled_data[0];
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audio_frame->data[1] = resampled_data[1];
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if (av_sample_fmt_is_planar(codec_context->sample_fmt)) {
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audio_frame->data[1] = resampled_data[1];
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}
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audio_frame->pts = frame_count * frame_size;
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audio_frame->pts = frame_count * frame_size;
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SendFrame(audio_frame.get());
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SendFrame(audio_frame.get());
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