Properly set #samples passed to encoder (#3204)

Summary:
Some audio encoders expect specific, exact number of samples described as in `AVCodecContext.frame_size`.

The `AVFrame.nb_samples` is set for the frames passed to `AVFilterGraph`,
but frames coming out of the graph do not necessarily have the same numbr of frames.

This causes issues with encoding OPUS (among others).

This commit fixes it by inserting `asetnsamples` to filter graph if a fixed number of samples is requested.

Note:
It turned out that FFmpeg 4.1 has issue with OPUS encoding. It does not properly discard some sample.
We should probably move the minimum required FFmpeg to 4.2, but I am not sure if we can enforce it via ABI.
Work around will be to issue an warning if encoding OPUS with 4.1. (follow-up)

Pull Request resolved: #3204

Reviewed By: nateanl

Differential Revision: D44374668

Pulled By: mthrok

fbshipit-source-id: 10ef5333dc0677dfb83c8e40b78edd8ded1b21dc

Author: mthrok

test/torchaudio_unittest/io/stream_writer_test.py

@@ -318,28 +318,58 @@ def test_video_num_frames(self, framerate, resolution, format):
             pass
 
     @nested_params(
-        ["wav", "mp3", "flac"],
+        ["wav", "flac"],
         [8000, 16000, 44100],
         [1, 2],
     )
-    def test_audio_num_frames(self, ext, sample_rate, num_channels):
-        """"""
+    def test_audio_num_frames_lossless(self, ext, sample_rate, num_channels):
+        """Lossless format preserves the data"""
         filename = f"test.{ext}"
 
+        data = get_sinusoid(sample_rate=sample_rate, n_channels=num_channels, dtype="int16", channels_first=False)
+
         # Write data
         dst = self.get_dst(filename)
         s = torchaudio.io.StreamWriter(dst=dst, format=ext)
-        s.add_audio_stream(sample_rate=sample_rate, num_channels=num_channels)
+        s.add_audio_stream(sample_rate=sample_rate, num_channels=num_channels, format="s16")
+        with s.open():
+            s.write_audio_chunk(0, data)
 
-        freq = 300
-        duration = 60
-        theta = torch.linspace(0, freq * 2 * 3.14 * duration, sample_rate * duration)
-        if num_channels == 1:
-            chunk = torch.sin(theta).unsqueeze(-1)
-        else:
-            chunk = torch.stack([torch.sin(theta), torch.cos(theta)], dim=-1)
+        if self.test_fileobj:
+            dst.flush()
+
+        # Load data
+        s = torchaudio.io.StreamReader(src=self.get_temp_path(filename))
+        s.add_audio_stream(-1)
+        s.process_all_packets()
+        (saved,) = s.pop_chunks()
+
+        self.assertEqual(saved, data)
+
+    @parameterized.expand(
+        [
+            ("mp3", 1, 8000),
+            ("mp3", 1, 16000),
+            ("mp3", 1, 44100),
+            ("mp3", 2, 8000),
+            ("mp3", 2, 16000),
+            ("mp3", 2, 44100),
+            ("opus", 1, 48000),
+        ]
+    )
+    def test_audio_num_frames_lossy(self, ext, num_channels, sample_rate):
+        """Saving audio preserves the number of channels and frames"""
+        filename = f"test.{ext}"
+
+        data = get_sinusoid(sample_rate=sample_rate, n_channels=num_channels, channels_first=False)
+
+        # Write data
+        dst = self.get_dst(filename)
+        s = torchaudio.io.StreamWriter(dst=dst, format=ext)
+        s.add_audio_stream(sample_rate=sample_rate, num_channels=num_channels)
         with s.open():
-            s.write_audio_chunk(0, chunk)
+            s.write_audio_chunk(0, data)
+
         if self.test_fileobj:
             dst.flush()
 
@@ -349,9 +379,21 @@ def test_audio_num_frames(self, ext, sample_rate, num_channels):
         s.process_all_packets()
         (saved,) = s.pop_chunks()
 
-        assert saved.shape == chunk.shape
-        if format in ["wav", "flac"]:
-            self.assertEqual(saved, chunk)
+        # This test fails for OPUS if FFmpeg is 4.1, but it passes for 4.2+
+        # 4.1 produces 48312 samples (extra 312)
+        # Probably this commit fixes it.
+        # https://github.com/FFmpeg/FFmpeg/commit/18aea7bdd96b320a40573bccabea56afeccdd91c
+        # TODO: issue warning if 4.1?
+        if ext == "opus":
+            ver = torchaudio.utils.ffmpeg_utils.get_versions()["libavcodec"]
+            # 5.1 libavcodec     59. 18.100
+            # 4.4 libavcodec     58.134.100
+            # 4.3 libavcodec     58. 91.100
+            # 4.2 libavcodec     58. 54.100
+            # 4.1 libavcodec     58. 35.100
+            if ver[0] < 59 and ver[1] < 54:
+                return
+        self.assertEqual(saved.shape, data.shape)
 
     def test_preserve_fps(self):
         """Decimal point frame rate is properly saved

torchaudio/csrc/ffmpeg/stream_writer/encode_process.cpp

@@ -233,10 +233,19 @@ FilterGraph get_audio_filter(
     AVCodecContext* codec_ctx) {
   auto desc = [&]() -> std::string {
     if (src_fmt == codec_ctx->sample_fmt) {
-      return "anull";
+      if (!codec_ctx->frame_size) {
+        return "anull";
+      } else {
+        std::stringstream ss;
+        ss << "asetnsamples=n=" << codec_ctx->frame_size << ":p=0";
+        return ss.str();
+      }
     } else {
       std::stringstream ss;
       ss << "aformat=" << av_get_sample_fmt_name(codec_ctx->sample_fmt);
+      if (codec_ctx->frame_size) {
+        ss << ",asetnsamples=n=" << codec_ctx->frame_size << ":p=0";
+      }
       return ss.str();
     }
   }();

torchaudio/csrc/ffmpeg/stream_writer/tensor_converter.cpp

@@ -40,9 +40,12 @@ void convert_func_(const torch::Tensor& chunk, AVFrame* buffer) {
   TORCH_INTERNAL_ASSERT_DEBUG_ONLY(chunk.dim() == 2);
   TORCH_INTERNAL_ASSERT_DEBUG_ONLY(chunk.size(1) == buffer->channels);
 
-  // TODO: make writable
   // https://ffmpeg.org/doxygen/4.1/muxing_8c_source.html#l00334
-  TORCH_CHECK(av_frame_is_writable(buffer), "frame is not writable.");
+  if (!av_frame_is_writable(buffer)) {
+    int ret = av_frame_make_writable(buffer);
+    TORCH_INTERNAL_ASSERT(
+        ret >= 0, "Failed to make frame writable: ", av_err2string(ret));
+  }
 
   auto byte_size = chunk.numel() * chunk.element_size();
   memcpy(buffer->data[0], chunk.data_ptr(), byte_size);