From a28eac08ae1fb3f381361eef5f5875ae57a5e8c7 Mon Sep 17 00:00:00 2001 From: zhupengfei Date: Thu, 27 Feb 2020 16:37:06 +0800 Subject: [PATCH] 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. --- src/core/dumping/ffmpeg_backend.cpp | 33 +++++++++++++---------------- 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/src/core/dumping/ffmpeg_backend.cpp b/src/core/dumping/ffmpeg_backend.cpp index 68b148bd3..5a81bed78 100644 --- a/src/core/dumping/ffmpeg_backend.cpp +++ b/src/core/dumping/ffmpeg_backend.cpp @@ -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 src_data = {reinterpret_cast(channel0.data()), reinterpret_cast(channel1.data())}; - std::array dst_data = {resampled_data[0] + sample_size * offset, - resampled_data[1] + sample_size * offset}; + + std::array 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]; - audio_frame->data[1] = resampled_data[1]; + 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]; - audio_frame->data[1] = resampled_data[1]; + 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());