summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--media/base/audio_buffer.cc18
-rw-r--r--media/base/audio_buffer_unittest.cc10
-rw-r--r--media/base/audio_decoder_config.cc2
-rw-r--r--media/base/audio_decoder_config.h3
-rw-r--r--media/base/sample_format.cc3
-rw-r--r--media/base/sample_format.h3
-rw-r--r--media/ffmpeg/ffmpeg_common.cc6
-rw-r--r--media/mojo/interfaces/media_types.mojom6
-rw-r--r--media/mojo/services/media_type_converters.cc1
9 files changed, 47 insertions, 5 deletions
diff --git a/media/base/audio_buffer.cc b/media/base/audio_buffer.cc
index 762e884..f118914 100644
--- a/media/base/audio_buffer.cc
+++ b/media/base/audio_buffer.cc
@@ -51,7 +51,8 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
return;
if (sample_format == kSampleFormatPlanarF32 ||
- sample_format == kSampleFormatPlanarS16) {
+ sample_format == kSampleFormatPlanarS16 ||
+ sample_format == kSampleFormatPlanarS32) {
// Planar data, so need to allocate buffer for each channel.
// Determine per channel data size, taking into account alignment.
int block_size_per_channel =
@@ -254,6 +255,12 @@ static inline int32 ConvertF32ToS32(float value) {
: value * std::numeric_limits<int32>::max());
}
+// No need for conversion. Return value as is. Keeping function to align with
+// code structure.
+static inline int32 ConvertS32ToS32(int32 value) {
+ return value;
+}
+
template <class Target, typename Converter>
void InterleaveToS32(const std::vector<uint8*>& channel_data,
size_t frames_to_copy,
@@ -322,6 +329,14 @@ void AudioBuffer::ReadFramesInterleavedS32(int frames_to_copy,
dest_data,
ConvertF32ToS32);
break;
+ case kSampleFormatPlanarS32:
+ // Format is planar signed 32 bit. Convert each value into int32 and
+ // insert into output channel data.
+ InterleaveToS32<int32>(channel_data_,
+ frames_to_copy,
+ trim_start_,
+ dest_data,
+ ConvertS32ToS32);
case kUnknownSampleFormat:
NOTREACHED();
break;
@@ -365,6 +380,7 @@ void AudioBuffer::TrimRange(int start, int end) {
switch (sample_format_) {
case kSampleFormatPlanarS16:
case kSampleFormatPlanarF32:
+ case kSampleFormatPlanarS32:
// Planar data must be shifted per channel.
for (int ch = 0; ch < channel_count_; ++ch) {
memmove(channel_data_[ch] + (trim_start_ + start) * bytes_per_channel,
diff --git a/media/base/audio_buffer_unittest.cc b/media/base/audio_buffer_unittest.cc
index 165161a..168445a 100644
--- a/media/base/audio_buffer_unittest.cc
+++ b/media/base/audio_buffer_unittest.cc
@@ -488,6 +488,16 @@ static scoped_refptr<AudioBuffer> MakeReadFramesInterleavedTestBuffer(
65536.0f / std::numeric_limits<int32>::max(),
frames,
base::TimeDelta::FromSeconds(0));
+ case kSampleFormatPlanarS32:
+ return MakeAudioBuffer<int32>(
+ sample_format,
+ channel_layout,
+ channel_count,
+ sample_rate,
+ 0.0f,
+ 65536.0f / std::numeric_limits<int32>::max(),
+ frames,
+ base::TimeDelta::FromSeconds(0));
case kSampleFormatU8:
case kUnknownSampleFormat:
EXPECT_FALSE(true);
diff --git a/media/base/audio_decoder_config.cc b/media/base/audio_decoder_config.cc
index d77d8da..4526859 100644
--- a/media/base/audio_decoder_config.cc
+++ b/media/base/audio_decoder_config.cc
@@ -150,6 +150,8 @@ std::string AudioDecoderConfig::GetHumanReadableCodecName() const {
return "pcm_mulaw";
case kCodecOpus:
return "opus";
+ case kCodecALAC:
+ return "alac";
}
NOTREACHED();
return "";
diff --git a/media/base/audio_decoder_config.h b/media/base/audio_decoder_config.h
index 199da84..9b2fe96 100644
--- a/media/base/audio_decoder_config.h
+++ b/media/base/audio_decoder_config.h
@@ -36,13 +36,14 @@ enum AudioCodec {
kCodecOpus = 12,
// kCodecEAC3 = 13,
kCodecPCM_ALAW = 14,
+ kCodecALAC = 15,
// DO NOT ADD RANDOM AUDIO CODECS!
//
// The only acceptable time to add a new codec is if there is production code
// that uses said codec in the same CL.
// Must always be equal to the largest entry ever logged.
- kAudioCodecMax = kCodecPCM_ALAW,
+ kAudioCodecMax = kCodecALAC,
};
// TODO(dalecurtis): FFmpeg API uses |bytes_per_channel| instead of
diff --git a/media/base/sample_format.cc b/media/base/sample_format.cc
index cf8f20f..464fc1b 100644
--- a/media/base/sample_format.cc
+++ b/media/base/sample_format.cc
@@ -20,6 +20,7 @@ int SampleFormatToBytesPerChannel(SampleFormat sample_format) {
case kSampleFormatS32:
case kSampleFormatF32:
case kSampleFormatPlanarF32:
+ case kSampleFormatPlanarS32:
return 4;
}
@@ -43,6 +44,8 @@ const char* SampleFormatToString(SampleFormat sample_format) {
return "Signed 16-bit planar";
case kSampleFormatPlanarF32:
return "Float 32-bit planar";
+ case kSampleFormatPlanarS32:
+ return "Signed 32-bit planar";
}
NOTREACHED() << "Invalid sample format provided: " << sample_format;
return "";
diff --git a/media/base/sample_format.h b/media/base/sample_format.h
index 7c3df70..7a48843 100644
--- a/media/base/sample_format.h
+++ b/media/base/sample_format.h
@@ -21,9 +21,10 @@ enum SampleFormat {
kSampleFormatF32, // Float 32-bit.
kSampleFormatPlanarS16, // Signed 16-bit planar.
kSampleFormatPlanarF32, // Float 32-bit planar.
+ kSampleFormatPlanarS32, // Signed 32-bit planar.
// Must always be equal to largest value ever logged.
- kSampleFormatMax = kSampleFormatPlanarF32,
+ kSampleFormatMax = kSampleFormatPlanarS32,
};
// Returns the number of bytes used per channel for the specified
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
index 2c99e7e..36d0e79 100644
--- a/media/ffmpeg/ffmpeg_common.cc
+++ b/media/ffmpeg/ffmpeg_common.cc
@@ -92,6 +92,8 @@ static AudioCodec CodecIDToAudioCodec(AVCodecID codec_id) {
return kCodecPCM_MULAW;
case AV_CODEC_ID_OPUS:
return kCodecOpus;
+ case AV_CODEC_ID_ALAC:
+ return kCodecALAC;
default:
DVLOG(1) << "Unknown audio CodecID: " << codec_id;
}
@@ -103,6 +105,8 @@ static AVCodecID AudioCodecToCodecID(AudioCodec audio_codec,
switch (audio_codec) {
case kCodecAAC:
return AV_CODEC_ID_AAC;
+ case kCodecALAC:
+ return AV_CODEC_ID_ALAC;
case kCodecMP3:
return AV_CODEC_ID_MP3;
case kCodecPCM:
@@ -242,6 +246,8 @@ SampleFormat AVSampleFormatToSampleFormat(AVSampleFormat sample_format) {
return kSampleFormatF32;
case AV_SAMPLE_FMT_S16P:
return kSampleFormatPlanarS16;
+ case AV_SAMPLE_FMT_S32P:
+ return kSampleFormatPlanarS32;
case AV_SAMPLE_FMT_FLTP:
return kSampleFormatPlanarF32;
default:
diff --git a/media/mojo/interfaces/media_types.mojom b/media/mojo/interfaces/media_types.mojom
index 032cb39..43db2b5 100644
--- a/media/mojo/interfaces/media_types.mojom
+++ b/media/mojo/interfaces/media_types.mojom
@@ -31,7 +31,8 @@ enum AudioCodec {
Opus = 12,
// EAC3 = 13,
PCM_ALAW = 14,
- MAX = PCM_ALAW,
+ ALAC = 15,
+ MAX = ALAC,
};
// See media/base/channel_layout.h for descriptions.
@@ -84,7 +85,8 @@ enum SampleFormat {
F32,
PlanarS16,
PlanarF32,
- Max = PlanarF32,
+ PlanarS32,
+ Max = PlanarS32,
};
// See media/base/video_frame.h for descriptions.
diff --git a/media/mojo/services/media_type_converters.cc b/media/mojo/services/media_type_converters.cc
index 51b9fda..a930716d 100644
--- a/media/mojo/services/media_type_converters.cc
+++ b/media/mojo/services/media_type_converters.cc
@@ -47,6 +47,7 @@ ASSERT_ENUM_EQ(AudioCodec, kCodec, AUDIO_CODEC_, PCM_S16BE);
ASSERT_ENUM_EQ(AudioCodec, kCodec, AUDIO_CODEC_, PCM_S24BE);
ASSERT_ENUM_EQ(AudioCodec, kCodec, AUDIO_CODEC_, Opus);
ASSERT_ENUM_EQ(AudioCodec, kCodec, AUDIO_CODEC_, PCM_ALAW);
+ASSERT_ENUM_EQ(AudioCodec, kCodec, AUDIO_CODEC_, ALAC);
ASSERT_ENUM_EQ_RAW(AudioCodec, kAudioCodecMax, AUDIO_CODEC_MAX);
// ChannelLayout.
generated by cgit v1.1 at 2025-01-24 21:59:02 +0000