diff --git a/src/audio_core/adsp/apps/opus/opus_decoder.cpp b/src/audio_core/adsp/apps/opus/opus_decoder.cpp
index 199ffe2f40..c163cd4077 100644
--- a/src/audio_core/adsp/apps/opus/opus_decoder.cpp
+++ b/src/audio_core/adsp/apps/opus/opus_decoder.cpp
@@ -94,6 +94,7 @@ public:
             avc->request_sample_fmt = AV_SAMPLE_FMT_S16;
             av_channel_layout_default(&avc->ch_layout, channel_count);
             if (avcodec_open2(avc, codec, nullptr) >= 0) {
+                ASSERT(avc->request_sample_fmt == AV_SAMPLE_FMT_S16);
                 pkt = av_packet_alloc();
                 frame = av_frame_alloc();
                 // opus fltp -> libopus s16
@@ -140,30 +141,27 @@ public:
             if ((r = av_new_packet(pkt, int(input_data_size))) >= 0) {
                 std::memcpy(pkt->data, reinterpret_cast<const u8*>(input_data), input_data_size);
                 if ((r = avcodec_send_packet(avc, pkt)) >= 0) {
-                    int rem_output_bytes = int(output_data_size);
                     av_frame_unref(frame);
-                    do {
-                        r = avcodec_receive_frame(avc, frame);
-                        auto const input_bsize = av_samples_get_buffer_size(frame->linesize, frame->ch_layout.nb_channels, frame->nb_samples, AV_SAMPLE_FMT_S16, 1);
-                        ASSERT(rem_output_bytes - input_bsize >= 0);
-                        auto const dst_data = reinterpret_cast<s16*>(output_data + (int(output_data_size) - rem_output_bytes));
-                        if (frame->format == AV_SAMPLE_FMT_S16) {
+                    while ((r = avcodec_receive_frame(avc, frame)) >= 0) {
+                        u8 *dst_arr[8] = {reinterpret_cast<u8*>(output_data), 0};
+                        if (frame->format == avc->request_sample_fmt) {
                             // input_bsize == output_bsize
-                            ASSERT(size_t(int(output_data_size) - rem_output_bytes + input_bsize) <= size_t(output_data_size));
-                            std::memcpy(dst_data, frame->data[0], size_t(input_bsize));
+                            av_samples_copy(dst_arr, frame->data, out_sample_count, 0, frame->nb_samples, frame->ch_layout.nb_channels, (enum AVSampleFormat)frame->format);
+                            out_sample_count += frame->nb_samples;
                         } else {
                             int out_samples = int(av_rescale_rnd(swr_get_delay(swr, frame->sample_rate) + frame->nb_samples, frame->sample_rate, frame->sample_rate, AV_ROUND_UP));
-                            uint8_t *output = nullptr;
-                            av_samples_alloc(&output, frame->linesize, frame->ch_layout.nb_channels, out_samples, AV_SAMPLE_FMT_S16, 0);
-                            out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)frame->data, frame->nb_samples);
-                            auto const output_bsize = av_samples_get_buffer_size(frame->linesize, frame->ch_layout.nb_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
-                            std::memcpy(dst_data, output, size_t(output_bsize));
-                            av_freep(&output);
+                            u8 *src_arr[8] = {0};
+                            if ((r = av_samples_alloc(src_arr, frame->linesize, frame->ch_layout.nb_channels, out_samples, avc->request_sample_fmt, 0)) >= 0) {
+                                out_samples = swr_convert(swr, src_arr, out_samples, (const u8 **)frame->data, frame->nb_samples);
+                                av_samples_copy(dst_arr, src_arr, out_sample_count, 0, out_samples, frame->ch_layout.nb_channels, avc->request_sample_fmt);
+                                av_freep(src_arr);
+                                out_sample_count += out_samples;
+                            } else {
+                                LOG_ERROR(Audio_DSP, "skipped");
+                            }
                         }
-                        out_sample_count += frame->nb_samples;
-                        rem_output_bytes -= input_bsize;
                         av_frame_unref(frame);
-                    } while (r >= 0 && rem_output_bytes > 0);
+                    }
                     if (r == AVERROR(EAGAIN) || r == AVERROR_EOF) {
                         // non-errors
                         LOG_DEBUG(Audio_DSP, "{}", r);
@@ -171,7 +169,6 @@ public:
                         // errors
                         LOG_ERROR(Audio_DSP, "{}", r);
                     }
-                    ASSERT(rem_output_bytes == 0 && "remaining bytes!");
                     av_packet_unref(pkt);
                     return ResultSuccess;
                 } else {