Refactor arg mapping in ffmpeg save function (#3387)

Summary:
The arguments of TorchAudio's save function ("format", "bits_per_sample" and "encoding")
are not one-to-one mapping to the arguments of FFmpeg encoding.

For example, to use vorbis codec, FFmpeg expects "ogg" container/extension with "vorbis"
encoder. It does not recognize "vorbis" extension like TorchAudio (libsox) does.

This commit refactors the logic to parse/map the arguments.

As a result it now properly works with vorbis and mp3 extension.

Pull Request resolved: https://github.com/pytorch/audio/pull/3387

Reviewed By: hwangjeff

Differential Revision: D46328787

Pulled By: mthrok

fbshipit-source-id: 36f993952a062bfec58a8b51be6aa86297571f90

test/torchaudio_unittest/backend/dispatcher/ffmpeg/load_test.py

@@ -7,7 +7,7 @@ from functools import partial
 from parameterized import parameterized
 from torchaudio._backend.utils import get_load_func
 from torchaudio._internal import module_utils as _mod_utils
-from torchaudio.io._compat import _get_encoder
+from torchaudio.io._compat import _parse_save_args
 
 from torchaudio_unittest.backend.dispatcher.sox.common import name_func
 from torchaudio_unittest.common_utils import (
@@ -56,11 +56,10 @@ class LoadTestBase(TempDirMixin, PytorchTestCase):
          |
          |    1. Generate given format with Sox
          |
-         v    3. Convert to wav with FFmpeg
-        given format ----------------------> wav
-         |                                   |
-         |    2. Load with torchaudio        | 4. Load with scipy
+         + ----------------------------------+ 3. Convert to wav with FFmpeg
          |                                   |
+         |    2. Load the given format       | 4. Load with scipy
+         |       with torchaudio             |
          v                                   v
         tensor ----------> x <----------- tensor
                        5. Compare
@@ -72,7 +71,6 @@ class LoadTestBase(TempDirMixin, PytorchTestCase):
         By combining i & ii, step 2. and 4. allow for loading reference given format
         data without using torchaudio
         """
-
         path = self.get_temp_path(f"1.original.{format}")
         ref_path = self.get_temp_path("2.reference.wav")
 
@@ -91,15 +89,15 @@ class LoadTestBase(TempDirMixin, PytorchTestCase):
 
         # 3. Convert to wav with ffmpeg
         if normalize:
-            acodec = "pcm_f32le"
+            encoder = "pcm_f32le"
         else:
             encoding_map = {
                 "floating-point": "PCM_F",
                 "signed-integer": "PCM_S",
                 "unsigned-integer": "PCM_U",
             }
-            acodec = _get_encoder(data.dtype, "wav", encoding_map.get(encoding), bit_depth)
-        _convert_audio_file(path, ref_path, acodec=acodec)
+            _, encoder, _ = _parse_save_args(format, format, encoding_map.get(encoding), bit_depth)
+        _convert_audio_file(path, ref_path, encoder=encoder)
 
         # 4. Load wav with scipy
         data_ref = load_wav(ref_path, normalize=normalize)[0]
@@ -277,7 +275,7 @@ class TestLoad(LoadTestBase):
         """`self._load` can load opus file correctly."""
         ops_path = get_asset_path("io", f"{bitrate}_{compression_level}_{num_channels}ch.opus")
         wav_path = self.get_temp_path(f"{bitrate}_{compression_level}_{num_channels}ch.opus.wav")
-        _convert_audio_file(ops_path, wav_path, acodec="pcm_f32le")
+        _convert_audio_file(ops_path, wav_path, encoder="pcm_f32le")
 
         expected, sample_rate = load_wav(wav_path)
         found, sr = self._load(ops_path)
@@ -301,15 +299,14 @@ class TestLoad(LoadTestBase):
     @parameterized.expand(
         list(
             itertools.product(
-                ["float32", "int32", "int16"],
-                [8000, 16000],
-                [1, 2],
+                ["int16"],
+                [3, 4, 16],
                 [False, True],
             )
         ),
         name_func=name_func,
     )
-    def test_amb(self, dtype, sample_rate, num_channels, normalize):
+    def test_amb(self, dtype, num_channels, normalize, sample_rate=8000):
         """`self._load` can load amb format correctly."""
         bit_depth = sox_utils.get_bit_depth(dtype)
         encoding = sox_utils.get_encoding(dtype)

test/torchaudio_unittest/backend/dispatcher/ffmpeg/save_test.py

@@ -8,7 +8,7 @@ from functools import partial
 import torch
 from parameterized import parameterized
 from torchaudio._backend.utils import get_save_func
-from torchaudio.io._compat import _get_encoder, _get_encoder_format
+from torchaudio.io._compat import _parse_save_args
 
 from torchaudio_unittest.backend.dispatcher.sox.common import get_enc_params, name_func
 from torchaudio_unittest.common_utils import (
@@ -24,12 +24,14 @@ from torchaudio_unittest.common_utils import (
 )
 
 
-def _convert_audio_file(src_path, dst_path, format=None, acodec=None):
-    command = ["ffmpeg", "-y", "-i", src_path, "-strict", "-2"]
-    if format:
-        command += ["-sample_fmt", format]
-    if acodec:
-        command += ["-acodec", acodec]
+def _convert_audio_file(src_path, dst_path, muxer=None, encoder=None, sample_fmt=None):
+    command = ["ffmpeg", "-hide_banner", "-y", "-i", src_path, "-strict", "-2"]
+    if muxer:
+        command += ["-f", muxer]
+    if encoder:
+        command += ["-acodec", encoder]
+    if sample_fmt:
+        command += ["-sample_fmt", sample_fmt]
     command += [dst_path]
     print(" ".join(command), file=sys.stderr)
     subprocess.run(command, check=True)
@@ -100,8 +102,10 @@ class SaveTestBase(TempDirMixin, TorchaudioTestCase):
         # 2.1. Convert the original wav to target format with torchaudio
         data = load_wav(src_path, normalize=False)[0]
         if test_mode == "path":
-            self._save(tgt_path, data, sample_rate, encoding=encoding, bits_per_sample=bits_per_sample)
+            ext = format
+            self._save(tgt_path, data, sample_rate, format=format, encoding=encoding, bits_per_sample=bits_per_sample)
         elif test_mode == "fileobj":
+            ext = None
             with open(tgt_path, "bw") as file_:
                 self._save(
                     file_,
@@ -113,6 +117,7 @@ class SaveTestBase(TempDirMixin, TorchaudioTestCase):
                 )
         elif test_mode == "bytesio":
             file_ = io.BytesIO()
+            ext = None
             self._save(
                 file_,
                 data,
@@ -127,16 +132,15 @@ class SaveTestBase(TempDirMixin, TorchaudioTestCase):
         else:
             raise ValueError(f"Unexpected test mode: {test_mode}")
         # 2.2. Convert the target format to wav with ffmpeg
-        _convert_audio_file(tgt_path, tst_path, acodec="pcm_f32le")
+        _convert_audio_file(tgt_path, tst_path, encoder="pcm_f32le")
         # 2.3. Load with SciPy
         found = load_wav(tst_path, normalize=False)[0]
 
         # 3.1. Convert the original wav to target format with ffmpeg
-        acodec = _get_encoder(data.dtype, format, encoding, bits_per_sample)
-        sample_fmt = _get_encoder_format(format, bits_per_sample)
-        _convert_audio_file(src_path, sox_path, acodec=acodec, format=sample_fmt)
+        muxer, encoder, sample_fmt = _parse_save_args(ext, format, encoding, bits_per_sample)
+        _convert_audio_file(src_path, sox_path, muxer=muxer, encoder=encoder, sample_fmt=sample_fmt)
         # 3.2. Convert the target format to wav with ffmpeg
-        _convert_audio_file(sox_path, ref_path, acodec="pcm_f32le")
+        _convert_audio_file(sox_path, ref_path, encoder="pcm_f32le")
         # 3.3. Load with SciPy
         expected = load_wav(ref_path, normalize=False)[0]
 

torchaudio/io/_compat.py

@@ -102,7 +102,8 @@ def load_audio_fileobj(
     format: Optional[str] = None,
     buffer_size: int = 4096,
 ) -> Tuple[torch.Tensor, int]:
-    s = torchaudio.lib._torchaudio_ffmpeg.StreamReaderFileObj(src, format, None, buffer_size)
+    demuxer = "ogg" if format == "vorbis" else format
+    s = torchaudio.lib._torchaudio_ffmpeg.StreamReaderFileObj(src, demuxer, None, buffer_size)
     sample_rate = int(s.get_src_stream_info(s.find_best_audio_stream()).sample_rate)
     filter = _get_load_filter(frame_offset, num_frames, convert)
     waveform = _load_audio_fileobj(s, filter, channels_first)
@@ -131,7 +132,7 @@ def _native_endianness() -> str:
         return "be"
 
 
-def _get_encoder_for_wav(dtype: torch.dtype, encoding: str, bits_per_sample: int) -> str:
+def _get_encoder_for_wav(encoding: str, bits_per_sample: int) -> str:
     if bits_per_sample not in {None, 8, 16, 24, 32, 64}:
         raise ValueError(f"Invalid bits_per_sample {bits_per_sample} for WAV encoding.")
     endianness = _native_endianness()
@@ -148,49 +149,93 @@ def _get_encoder_for_wav(dtype: torch.dtype, encoding: str, bits_per_sample: int
         if bits_per_sample == 8:
             raise ValueError("For WAV signed PCM, 8-bit encoding is not supported.")
         return f"pcm_s{bits_per_sample}{endianness}"
-    elif encoding == "PCM_U":
+    if encoding == "PCM_U":
         if bits_per_sample in (None, 8):
             return "pcm_u8"
         raise ValueError("For WAV unsigned PCM, only 8-bit encoding is supported.")
-    elif encoding == "PCM_F":
+    if encoding == "PCM_F":
         if not bits_per_sample:
             bits_per_sample = 32
         if bits_per_sample in (32, 64):
             return f"pcm_f{bits_per_sample}{endianness}"
         raise ValueError("For WAV float PCM, only 32- and 64-bit encodings are supported.")
-    elif encoding == "ULAW":
+    if encoding == "ULAW":
         if bits_per_sample in (None, 8):
             return "pcm_mulaw"
         raise ValueError("For WAV PCM mu-law, only 8-bit encoding is supported.")
-    elif encoding == "ALAW":
+    if encoding == "ALAW":
         if bits_per_sample in (None, 8):
             return "pcm_alaw"
         raise ValueError("For WAV PCM A-law, only 8-bit encoding is supported.")
     raise ValueError(f"WAV encoding {encoding} is not supported.")
 
 
-def _get_encoder(dtype: torch.dtype, format: str, encoding: str, bits_per_sample: int) -> str:
-    if format == "wav":
-        return _get_encoder_for_wav(dtype, encoding, bits_per_sample)
-    if format == "flac":
-        return "flac"
-    if format in ("ogg", "vorbis"):
-        if encoding or bits_per_sample:
-            raise ValueError("ogg/vorbis does not support encoding/bits_per_sample.")
-        return "vorbis"
-    return format
+def _get_flac_sample_fmt(bps):
+    if bps is None or bps == 16:
+        return "s16"
+    if bps == 24:
+        return "s32"
+    raise ValueError(f"FLAC only supports bits_per_sample values of 16 and 24 ({bps} specified).")
 
 
-def _get_encoder_format(format: str, bits_per_sample: Optional[int]) -> str:
-    if format == "flac":
-        if not bits_per_sample:
-            return "s16"
-        if bits_per_sample == 24:
-            return "s32"
-        if bits_per_sample == 16:
-            return "s16"
-        raise ValueError(f"FLAC only supports bits_per_sample values of 16 and 24 ({bits_per_sample} specified).")
-    return None
+def _parse_save_args(
+    ext: Optional[str],
+    format: Optional[str],
+    encoding: Optional[str],
+    bps: Optional[int],
+):
+    # torchaudio's save function accepts the followings, which do not 1to1 map
+    # to FFmpeg.
+    #
+    # - format: audio format
+    # - bits_per_sample: encoder sample format
+    # - encoding: such as PCM_U8.
+    #
+    # In FFmpeg, format is specified with the following three (and more)
+    #
+    # - muxer: could be audio format or container format.
+    # the one we passed to the constructor of StreamWriter
+    # - encoder: the audio encoder used to encode audio
+    # - encoder sample format: the format used by encoder to encode audio.
+    #
+    # If encoder sample format is different from source sample format, StreamWriter
+    # will insert a filter automatically.
+    #
+    def _type(spec):
+        # either format is exactly the specified one
+        # or extension matches to the spec AND there is no format override.
+        return format == spec or (format is None and ext == spec)
+
+    if _type("wav") or _type("amb"):
+        # wav is special because it supports different encoding through encoders
+        # each encoder only supports one encoder format
+        #
+        # amb format is a special case originated from libsox.
+        # It is basically a WAV format, with slight modification.
+        # https://github.com/chirlu/sox/commit/4a4ea33edbca5972a1ed8933cc3512c7302fa67a#diff-39171191a858add9df87f5f210a34a776ac2c026842ae6db6ce97f5e68836795
+        # It is a format so that decoders will recognize it as ambisonic.
+        # https://www.ambisonia.com/Members/mleese/file-format-for-b-format/
+        # FFmpeg does not recognize amb because it is basically a WAV format.
+        muxer = "wav"
+        encoder = _get_encoder_for_wav(encoding, bps)
+        sample_fmt = None
+    elif _type("vorbis"):
+        # FFpmeg does not recognize vorbis extension, while libsox used to do.
+        # For the sake of bakward compatibility, (and the simplicity),
+        # we support the case where users want to do save("foo.vorbis")
+        muxer = "ogg"
+        encoder = "vorbis"
+        sample_fmt = None
+    else:
+        muxer = format
+        encoder = None
+        sample_fmt = None
+        if _type("flac"):
+            sample_fmt = _get_flac_sample_fmt(bps)
+        if _type("ogg"):
+            sample_fmt = _get_flac_sample_fmt(bps)
+    print(ext, format, encoding, bps, "===>", muxer, encoder, sample_fmt)
+    return muxer, encoder, sample_fmt
 
 
 # NOTE: in contrast to load_audio* and info_audio*, this function is NOT compatible with TorchScript.
@@ -204,25 +249,27 @@ def save_audio(
     bits_per_sample: Optional[int] = None,
     buffer_size: int = 4096,
 ) -> None:
+    ext = None
     if hasattr(uri, "write"):
         if format is None:
             raise RuntimeError("'format' is required when saving to file object.")
     else:
         uri = os.path.normpath(uri)
-    s = StreamWriter(uri, format=format, buffer_size=buffer_size)
-    if format is None:
-        tokens = str(uri).split(".")
-        if len(tokens) > 1:
-            format = tokens[-1].lower()
+        if tokens := str(uri).split(".")[1:]:
+            ext = tokens[-1].lower()
+
+    muxer, encoder, enc_fmt = _parse_save_args(ext, format, encoding, bits_per_sample)
 
     if channels_first:
         src = src.T
+
+    s = StreamWriter(uri, format=muxer, buffer_size=buffer_size)
     s.add_audio_stream(
         sample_rate,
         num_channels=src.size(-1),
         format=_get_sample_format(src.dtype),
-        encoder=_get_encoder(src.dtype, format, encoding, bits_per_sample),
-        encoder_format=_get_encoder_format(format, bits_per_sample),
+        encoder=encoder,
+        encoder_format=enc_fmt,
     )
     with s.open():
         s.write_audio_chunk(0, src)