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Unified Diff: media/base/audio_buffer.cc

Issue 600143002: Adding new function ReadFrames() that returns the audio frame in planar (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Adding trim_start_. Created 6 years, 3 months ago
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Index: media/base/audio_buffer.cc
diff --git a/media/base/audio_buffer.cc b/media/base/audio_buffer.cc
index 33d4ecbb1abbd4b3bdc709bdd3a817f530d872a9..fcc35044f376400d978fe4aa46437159908fd1e1 100644
--- a/media/base/audio_buffer.cc
+++ b/media/base/audio_buffer.cc
@@ -243,6 +243,89 @@ void AudioBuffer::ReadFrames(int frames_to_copy,
source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
}
+static const int32 kUint8Bias = 128;
+
+static inline int32 ConvertU8ToS32(uint8 value) {
+ return (static_cast<int32>(value) - kUint8Bias) << 24;
+}
+
+static inline int32 ConvertS16ToS32(int16 value) {
+ return static_cast<int32>(value) << 16;
+}
+
+static inline int32 ConvertF32ToS32(float value) {
+ return static_cast<int32>(value < 0
+ ? value * ~std::numeric_limits<int32>::min()
DaleCurtis 2014/09/25 22:30:44 I don't think this does what you want; e.g., ~(-5)
+ : value * std::numeric_limits<int32>::max());
+}
+
+void AudioBuffer::ReadFramesInterleavedS32(int frames_to_copy,
+ int32* dest_data) {
+ DCHECK_EQ(frames_to_copy, channel_count_ * adjusted_frame_count_);
+
+ if (sample_format_ == kSampleFormatU8) {
+ // Unsigned 8-bit w/ bias of 128.
+ const uint8* source_data =
+ reinterpret_cast<const uint8*>(channel_data_[0]) + trim_start_;
+
+ for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+ dest_data[i] = ConvertU8ToS32(source_data[i]);
+ }
+ } else if (sample_format_ == kSampleFormatS16) {
+ // Format is interleaved signed16. Convert each value into int32 and insert
+ // into output channel data.
+ const int16* source_data =
+ reinterpret_cast<const int16*>(channel_data_[0]) + trim_start_;
+
+ for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+ dest_data[i] = ConvertS16ToS32(source_data[i]);
+ }
+ } else if (sample_format_ == kSampleFormatS32) {
+ // Format is interleaved signed32; just copy the data.
+ const int32* source_data =
+ reinterpret_cast<const int32*>(channel_data_[0]) + trim_start_;
+ memcpy(dest_data,
+ source_data,
+ adjusted_frame_count_ * channel_count_ * sizeof(int32));
+ } else if (sample_format_ == kSampleFormatF32) {
+ // Format is interleaved float. Convert each value into int32 and insert
+ // into output channel data.
+ const float* source_data =
+ reinterpret_cast<const float*>(channel_data_[0]) + trim_start_;
+ for (int i = 0; i < adjusted_frame_count_ * channel_count_; ++i) {
+ dest_data[i] = ConvertF32ToS32(source_data[i]);
+ }
+ } else if (sample_format_ == kSampleFormatPlanarS16) {
+ // Format is planar signed 16 bit. Convert each value into int32 and insert
+ // into output channel data.
+ int dest_data_frame_size = channel_count_;
+ for (int ch = 0; ch < channel_count_; ++ch) {
+ const int16* source_data =
+ reinterpret_cast<const int16*>(channel_data_[ch]) + trim_start_;
+
+ for (int i = 0; i < adjusted_frame_count_; ++i) {
+ dest_data[ch + (i * dest_data_frame_size)] =
DaleCurtis 2014/09/25 22:30:44 A paired incrementing offset is more efficient tha
+ ConvertS16ToS32(source_data[i]);
+ }
+ }
+ } else if (sample_format_ == kSampleFormatPlanarF32) {
+ // Format is planar float. Convert each value into int32 and insert into
+ // output channel data.
+ int dest_data_frame_size = channel_count_;
+ for (int ch = 0; ch < channel_count_; ++ch) {
+ const float* source_data =
+ reinterpret_cast<const float*>(channel_data_[ch]) + trim_start_;
+
+ for (int i = 0; i < adjusted_frame_count_; ++i) {
+ dest_data[ch + (i * dest_data_frame_size)] =
DaleCurtis 2014/09/25 22:30:44 Ditto.
+ ConvertF32ToS32(source_data[i]);
+ }
+ }
+ } else {
+ NOTREACHED();
DaleCurtis 2014/09/25 22:30:44 If you prefer you could use case statement since i
+ }
+}
+
void AudioBuffer::TrimStart(int frames_to_trim) {
CHECK_GE(frames_to_trim, 0);
CHECK_LE(frames_to_trim, adjusted_frame_count_);
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