rename 16ch file, remove <8 channels check

Author: Dan-Flores

src/torchcodec/_core/SingleStreamDecoder.cpp

@@ -571,9 +571,8 @@ void SingleStreamDecoder::addAudioStream(
       "seek_mode must be 'approximate' for audio streams.");
   if (audioStreamOptions.numChannels.has_value()) {
     TORCH_CHECK(
-        *audioStreamOptions.numChannels > 0 &&
-            *audioStreamOptions.numChannels <= AV_NUM_DATA_POINTERS,
-        "num_channels must be > 0 and <= AV_NUM_DATA_POINTERS (usually 8). Got: ",
+        *audioStreamOptions.numChannels > 0,
+        "num_channels must be > 0. Got: ",
         *audioStreamOptions.numChannels);
   }
 

test/resources/16ch.wav test/resources/sine_16ch_s16.wavtest/resources/16ch.wav renamed to test/resources/sine_16ch_s16.wav

@@ -43,7 +43,7 @@
     needs_cuda,
     needs_ffmpeg_cli,
     psnr,
-    SINE_16_CHANNELS,
+    SINE_16_CHANNEL_S16,
     SINE_MONO_S16,
     SINE_MONO_S32,
     SINE_MONO_S32_44100,
@@ -1791,7 +1791,9 @@ def test_cpu_fallback_no_fallback_on_supported_video(self, device):
 
 
 class TestAudioDecoder:
-    @pytest.mark.parametrize("asset", (NASA_AUDIO, NASA_AUDIO_MP3, SINE_MONO_S32))
+    @pytest.mark.parametrize(
+        "asset", (NASA_AUDIO, NASA_AUDIO_MP3, SINE_MONO_S32, SINE_16_CHANNEL_S16)
+    )
     def test_metadata(self, asset):
         decoder = AudioDecoder(asset.path)
         assert isinstance(decoder.metadata, AudioStreamMetadata)
@@ -1849,7 +1851,7 @@ def test_get_all_samples_with_range(self, asset, stop_seconds):
         assert samples.sample_rate == asset.sample_rate
         assert samples.pts_seconds == asset.get_frame_info(idx=0).pts_seconds
 
-    @pytest.mark.parametrize("asset", (NASA_AUDIO, NASA_AUDIO_MP3, SINE_16_CHANNELS))
+    @pytest.mark.parametrize("asset", (NASA_AUDIO, NASA_AUDIO_MP3, SINE_16_CHANNEL_S16))
     def test_get_all_samples(self, asset):
         decoder = AudioDecoder(asset.path)
         torch.testing.assert_close(
@@ -2118,8 +2120,7 @@ def test_samples_duration(self, asset, sample_rate):
     # that the extra tensor allocation that happens within
     # maybeFlushSwrBuffers() is correct.
     @pytest.mark.parametrize("sample_rate", (None, 16_000))
-    # FFmpeg can handle up to AV_NUM_DATA_POINTERS=8 channels
-    @pytest.mark.parametrize("num_channels", (1, 2, 8, None))
+    @pytest.mark.parametrize("num_channels", (1, 2, 8, 16, None))
     def test_num_channels(self, asset, sample_rate, num_channels):
         decoder = AudioDecoder(
             asset.path, sample_rate=sample_rate, num_channels=num_channels
@@ -2133,12 +2134,5 @@ def test_num_channels(self, asset, sample_rate, num_channels):
 
     @pytest.mark.parametrize("asset", (SINE_MONO_S32, NASA_AUDIO_MP3))
     def test_num_channels_errors(self, asset):
-        with pytest.raises(
-            RuntimeError, match="num_channels must be > 0 and <= AV_NUM_DATA_POINTERS"
-        ):
+        with pytest.raises(RuntimeError, match="num_channels must be > 0"):
             AudioDecoder(asset.path, num_channels=0)
-        with pytest.raises(
-            RuntimeError, match="num_channels must be > 0 and <= AV_NUM_DATA_POINTERS"
-        ):
-            # FFmpeg can handle up to AV_NUM_DATA_POINTERS=8 channels
-            AudioDecoder(asset.path, num_channels=9)

…es/16ch.wav.stream0.all_frames_info.json …6ch_s16.wav.stream0.all_frames_info.jsontest/resources/16ch.wav.stream0.all_frames_info.json renamed to test/resources/sine_16ch_s16.wav.stream0.all_frames_info.json test/resources/16ch.wav.stream0.all_frames_info.json renamed to test/resources/sine_16ch_s16.wav.stream0.all_frames_info.json

@@ -836,9 +836,9 @@ def sample_format(self) -> str:
 )
 
 # 16-channel audio for testing support for >8 channels. Generated with:
-# ffmpeg -i test/resources/sine_mono_s32.wav -t 1 -filter_complex "[0]asplit=16[s0][s1][s2][s3][s4][s5][s6][s7][s8][s9][s10][s11][s12][s13][s14][s15];[s0][s1][s2][s3][s4][s5][s6][s7][s8][s9][s10][s11][s12][s13][s14][s15]amerge=inputs=16" -c:a pcm_s16le test/resources/16ch.wav
-SINE_16_CHANNELS = TestAudio(
-    filename="16ch.wav",
+# ffmpeg -i test/resources/sine_mono_s32.wav -t 1 -filter_complex "[0]asplit=16[s0][s1][s2][s3][s4][s5][s6][s7][s8][s9][s10][s11][s12][s13][s14][s15];[s0][s1][s2][s3][s4][s5][s6][s7][s8][s9][s10][s11][s12][s13][s14][s15]amerge=inputs=16" -c:a pcm_s16le test/resources/sine_16ch_s16.wav
+SINE_16_CHANNEL_S16 = TestAudio(
+    filename="sine_16ch_s16.wav",
     default_stream_index=0,
     frames={},  # Automatically loaded from json file
     stream_infos={